I believe by spec its 2* the number of physical channels, but don't
quote me on that. Specifically thats for everything going on on those
spans including suspended calls (which we don't currently support
anyway)
Mike
On Apr 9, 2008, at 4:08 PM, Andy Spitzer wrote:
Woof!
On Wed, 09 Apr
error LNK1181: cannot open input file
'..\..\..\..\w32\library\debug\freeswitchcore.lib'
Regards,
Tamas
Michael Jerris írta:
What os/compiler versions?
Mike
On Apr 9, 2008, at 2:16 PM, Tamas Cseke wrote:
Hello,
I have 2 problems:
1, libvoipcodecs
C2143: syntax error : missing
that that is different between our environments. I will fix up the
inet_pton issues tomorrow (later today) and let you know.
Mike
On Apr 10, 2008, at 1:57 AM, Tamas wrote:
Hi,
As I wrote in previous mail, it was svn trunk r8070.
Regards,
Tamas
Michael Jerris írta:
Is this with svn trunk (what
We have not added any MSC type functionality at this time.
Mike
On Apr 11, 2008, at 2:42 PM, Anya wrote:
Hi all,
can someone tell me if this project supports IuUP interface? I am
mainly
interested in support for Iu-CS.
I am looking for a small MSC that supports A and IuUP interfaces
I think you will need to try to tweak the build for unicode support to
fix this.
Mike
On Apr 15, 2008, at 11:48 AM, Jonas Gauffin wrote:
assert is made on while (*s *s != '' (*s != '%' || t != '%')
!isspace((int) (*s)))
*s contains ösvan
Full stack trace:
I think with the volume of calls you are handling, this is one place
where openser will serve you better than freeswitch. You said you
already have openser in this role, why would you not want to use it?
Mike.
On Apr 16, 2008, at 1:09 PM, kokoska rokoska [EMAIL PROTECTED]
wrote:
That being said, there is an open bug that the tarballs only work with
automake 1.9. That will be fixed in the next rc.
Mike
On Apr 16, 2008, at 10:54 PM, Anthony Minessale [EMAIL PROTECTED]
wrote:
rc2 and up is pre bootstrapped.
On Wed, Apr 16, 2008 at 10:46 PM, Peder @
DOH!! I knew I forgot something. Will roll those this morning. You
can grab the rc2 ones, I think there was only a couple small changes.
On Apr 17, 2008, at 6:55 AM, Peder @ NetworkOblivion [EMAIL PROTECTED]
wrote:
OK, I got through the make and install with rc3, but now I am trying
We don't have any real sizing numbers on this, but I would guess that
having your location for recorded messages on ram disk would make a
big difference. A bit of sipp magic with pcap playback will probably
give you some real world numbers and we would love to hear how much
you can scale.
Typically sip handsets have a reject button. If there is not a button
for this on your handset, then there is no way to do it from that
handset.
Mike
On May 19, 2008, at 9:34 AM, Czaderna wrote:
Hello
I've got question about rejecting a calls. I make a call between two
wireless phones
As stated before, this is the WRONG way to set this up. Please setup
sangoma as TDM-api and use span_wanpipe.
Mike
On May 21, 2008, at 10:13 AM, Helmut Kuper wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
hm, I still can't find a reproducable way for setting up FS with
I have done a very basic msi that visual studio builds, but it does
not include bootstrapper for the runtime or the sound files. If
someone has good experience with doing a more full installer, I would
love to have them take a look.
Mike
On May 21, 2008, at 1:53 PM, jeff sacksteder wrote:
We fixed this in make sofia-reconf late yesterday. Please update and
try again
On May 22, 2008, at 12:38 PM, RR wrote:
On Sat, May 17, 2008 at 12:29 PM, Brian West [EMAIL PROTECTED]
wrote:
This is a requirement if you're following trunk:
svn update
make sofia-reconf current vm-sync
Hi
.
Updated to revision 8528.
after this make sofia-reconf current vm-sync still gives the same
error. Same is the case on make current
Thx
On Thu, May 22, 2008 at 1:27 PM, Michael Jerris [EMAIL PROTECTED]
wrote:
We fixed this in make sofia-reconf late yesterday. Please update
and try again
Currently the directory interface is only used for that dialplan, I
would like to enhance that in the future. The directory dialploan
uses a filter of exten=destination number, and then has name/value
pairs, I will see if I can find the schema we used back when we
developed it, short of
Is this freeswitch on both sides of the call here? Can i get into the
bod live to trouble shoot this?
Mike
On Jun 1, 2008, at 6:35 AM, UV wrote:
I've checked, double checked and triple checked. Both configurations
are
identical.
Take a look at the pastebin capture:
at 8:09 AM, Michael Jerris [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Currently the directory interface is only used for that
dialplan, I
would like to enhance that in the future. The directory
dialploan
uses a filter of exten=destination number, and then has name
I reviewed your bug on jira. Your mera is sending a 500 error.
Please look into this problem with whoever provides support for your
mera.
Mike
On Jun 2, 2008, at 12:59 AM, Pieter Eduard wrote:
Hey all,
thanks for the reply, appreciate it so much,, my fs version is
FreeSwitch Version
It does not at this time. We have looked at adding support for rtcp
when the standards for doing rtcp on the same ports as rtp are
solidified.
Mike
On Jun 2, 2008, at 8:29 AM, Peder @ NetworkOblivion wrote:
Does FS support RTCP? I am interested in getting per call quality
stats
from
.
Let me know how can I help troubleshoot it.
If there's a way I can disable the Allow-Events headers, I'll be
able to
confirm my hypothesis.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Michael
Jerris
Sent: Monday, June 02, 2008 3:15 AM
Contact me offlist and can we arrange for me to get into your box
remotely.
Mike
On Jun 13, 2008, at 8:01 PM, Areski K wrote:
Anyone have been working with an MSSQL ODBC connection through
Javascript FS !?
I am facing this constant error :
2008-06-14 01:38:22 [CRIT] switch_odbc.c:240
Any bay area freeswith users interested in catching up for a drink
sometime this week, please drop me an e-mail off list.
Mike
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There were problems with it in all of the rc's but it should be fine
in 1.0.0
Mike
On Jun 18, 2008, at 6:44 PM, UV [EMAIL PROTECTED] wrote:
It’s quite embarrassing but I read about all the great things mod_xm
l_rpc can do, see all the threads about it, even manage to load the
module
I believe it does, we should re-try the send in this case. Can you
file a bug for this on jira (with a patch would make it even better :D )
On Jun 19, 2008, at 12:43 PM, Jonas Gauffin wrote:
Hello
I got some problems with the event socket, I do not receive all
events.
I've confirmed
If you want to actually not send to them you need to set the effective
caller id name/number.
Mike
On Jun 20, 2008, at 1:10 PM, Brian West wrote:
It will always contain the number. Its the far ends responsibility to
honor the privacy flags. The same happens on a PRI as far as I have
You can likely do this with a combination of broadcast (see
uuid_broadcast api command) and mod_tone_stream which lets you use
teletone just like you are playing a file.
Mike
On Jun 22, 2008, at 10:49 PM, Klaus Teller wrote:
Hi Folks,
I was was wondering if it's possible to play a
If your not using sip auth there is no reason to use a gateway.
Mike
On Jun 26, 2008, at 8:12 AM, Nick Temple wrote:
Hello,
I'm attempting to get Mix Meeting work with freeswitch they use
IP authorization only, however username / password are required in
the config file.
If I
Conferences are started dynamically when the first participant enters,
and end when there are no participants. There is no need to start a
conference before any participants enter.
Mike
On Jun 26, 2008, at 3:24 PM, e schmidbauer wrote:
is there anyway to start a conference automaticaly
Have you checked out http://wiki.freeswitch.org ?
Mike
On Jun 27, 2008, at 12:10 PM, Arthur wrote:
I personaly don't see the use of XML as a problem, but I see a lack
of documentation about the whole FS project as a problem, because
when someone doesn't understand how to manage a FS server
It is an rfc violation to start at 0. Thats a bug in your other
device. In regards to the volume, it should not matter, we set it
based on what we had seen on a lot of other devices, we should
probably make that pass through in the same way we do duration. I
doubt either of these is the
On Jun 29, 2008, at 12:08 PM, Henk Oegema wrote:
On Sunday 29 June 2008 17:44:29 Михаил Кривушин wrote:
В сообщении от 29 июня 2008 вы написали:
HO On Sunday 29 June 2008 15:38:25 Михаил Кривушин wrote:
HO Are you try to read log file? Can you post lines with error?
HO
HO
HO Many lines
You can add users (people who register to you) but not gateways
without restarting the sip profile.
On Jul 2, 2008, at 3:14 AM, Anton wrote:
Sorry, just a little not sure to understand correctly -
there is no way to add a new SIP account without restarting
the SIP profile, and so any such
Most likely its not actually matching the extension or it runs out of
actions to perform, can you post the full debug logs from the console?
Mike
On Jul 2, 2008, at 1:14 PM, Hristo Benev wrote:
Оригинално писмо
От: Michael Jerris
Относно: Re: [Freeswitch-users] How
is obfuscated
Оригинално писмо
От: Michael Jerris
Относно: Re: [Freeswitch-users] How to Configure SIP DID to IVR
До: freeswitch-users@lists.freeswitch.org
Изпратено на: Сряда, 2008, Юли 2 20:22:31 EEST
Most likely its not actually matching the extension or it runs out
Transfer?
On Jul 3, 2008, at 9:18 AM, Ghulam Mustafa wrote:
i never want to call a sip endpoint, all i want to do is pop a call
from queue and bridge it with a call group, and i want to accomplish
all
of this using JS.
Regards,
Mustafa
On Thu, 2008-07-03 at 08:59 -0400, Johny
I doubt we could embed erlang (as I have never seen it done) but it
would most likely be able to control freeswitch over a socket.
Mike
On Jul 3, 2008, at 10:19 AM, Kre?imir Tonkovi? wrote:
Hi!
I myself have bumbed into performance problems with heavy-weight
languages like python and
There are very few differences in windows, all the examples apply to
all operating systems.
Mike
On Jul 7, 2008, at 4:41 PM, Kin Quek wrote:
Hi All,
I finally got FS compiled on Windows Vista (with Visual Studio 2008
Express). I am ready to give it a test run. Since I am a newbie to
Can you please attach this patch as a file to a bug on
jira.freeswitch.org?
Mike
On Jul 7, 2008, at 5:36 PM, Chris Danielson wrote:
FreeSWITCH may or may not need this but I added an auto-record
feature into the mod_conference module. Essentially, one can
specify within the
someone was worried that their xml parser would not maintain order, so
we added that to specify in data the order of the callflows.
Mike
On Jul 9, 2008, at 11:19 AM, Michael S Collins wrote:
What is this value for? Does it relate to the callflow
profile_index=1 section of the XML cdr?
What do you get now when you try to do this?
Mike
On Jul 9, 2008, at 10:58 AM, Andy Spitzer wrote:
Woof!
On Tue, 08 Jul 2008 19:13:24 -0400, Anthony Minessale [EMAIL PROTECTED]
wrote:
try now =D
Ahh, now there is joy!
Can I trouble you to also make tone_stream://%(150, 0, 0) (or
We don't have one in tree, but it wouldn't be hard to write one.
Mike
On Jul 9, 2008, at 5:20 PM, Michael Collins wrote:
Followup - wait about good ol' fashioned silence detection, akin to
WaitForSilence in ast?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:freeswitch-
There is no list that I know of. This means there is no user [EMAIL PROTECTED]
in the user directory.
Mike
On Jul 10, 2008, at 11:49 AM, Jair Santos wrote:
I've been dealing since yesterday with the message
2008-07-10 08:44:10 [WARNING] sofia_reg.c:1061
sofia_reg_parse_auth() can't
There are definately fixes in XML CDR module in trunk that you will
want. Please update and let us know if it persists.
Mike
On Jul 11, 2008, at 9:55 AM, Faraz R. Khan [EMAIL PROTECTED]
wrote:
Core dumped again. With debug logging on the error happens in
mod_xml_cdr looks like :
Isnt it --without-libcurl ?
On Jul 11, 2008, at 10:13 AM, Brian West [EMAIL PROTECTED] wrote:
You're using curl to post cdr it looks like. Are you using your
system curl or the one we ship?
reconfigure freeswitch with this flag --without-curl
The system curl must have been used and has
Check out:
http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation
just setting the group_confirm_file without setting group_confirm_key
I think will do what you want.
Mike
On Jul 13, 2008, at 12:31 PM, Adnan Barakat wrote:
Hi All,
I'm looking for a way to play an
You are correct, You should be able to use:
http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#exec_in_answer_confirm
or
http://wiki.freeswitch.org/wiki/Channel_Variables#execute_on_answer
using playback.
Mike
On Jul 14, 2008, at 10:18 AM, Adnan Barakat wrote:
Michael Jerris wrote
)
all the best.
2008/7/14 Michael Jerris [EMAIL PROTECTED]:
I tried and got things this far but ran in to some linking weirdness
if I recall that caused this problem and I gave up. We are happy to
take patches to add openbsd support but we have no plans to fix it
ourselves.
Mike
On Jul
Open bsd does not support static linking? thats the problem... its
not properly linking the modules.. in foact, it should not even need
to link that in here... just to the core
Mike
*** Warning: This system can not link to static lib archive /usr/src/
On Jul 14, 2008, at 9:18 PM, Jair Santos wrote:
I am trying again this message.
Hi all,
I've created the following internal2.xml in the sip_profiles in
order to
register a phone outside the network (NAT involved).
I am getting Registration error 403 forbidden in the phone and
http://wiki.freeswitch.org/wiki/Mod_commands#chat
Usage: chat,proto|from|to|message,chat
i.e. chat sip [EMAIL PROTECTED] [EMAIL PROTECTED] this is a test
Mike
p.s. this command currently will work with sip, jabber (dingaling) and
iax.
On Jul 15, 2008, at 1:42 AM, David Knell wrote:
Hi -
You could also use the inband dtmf generator and queue_dtmf.
Mike
On Jul 15, 2008, at 9:45 AM, Anthony Minessale wrote:
there is no way to do both inband and info/2833 from within sip.
inband dtmf is not part of the SIP module it's part of the core.
you can generate an inband dtmf stream
The analysis looks good, but how in the API do we know when the
response is complete? Can you please open a bug on
jira.freeswitch.org for this so we can track it.
Thanks
Mike
On Jul 15, 2008, at 8:12 PM, Simon Tang [EMAIL PROTECTED] wrote:
Hello,
I’ve been making Javascript curl
Registration is not required, you have it configured however to
require authentication. You need to either provide credentials for
the gateway to authenticate with or have the calls come to a profile
that does not require auth.
Mike
On Jul 16, 2008, at 11:31 AM, whudson05 wrote:
Hi,
Check out scripts/socket in the source tree for a few examples. There
are some others in scripts/contrib as well.
Mike
On Jul 16, 2008, at 8:14 PM, [EMAIL PROTECTED] wrote:
On the FreeSWITCH wiki page entitled How does FreeSWITCH compare to
Asterisk? (http://www.freeswitch.org/node/117)
it should in bypass_media or proxy_media modes. in the other modes we
are in the media path and would not know how to handle the encrypted
packets.
Mike
On Jul 17, 2008, at 5:11 PM, Peter P GMX wrote:
Hello,
did anybody get Twinkle with ZRTPworking?
I tried this with 2 Twinkle clients
Are those your real user id and password?
Mike
On Jul 18, 2008, at 11:12 PM, Matt Darnell wrote:
If this is the real username and secret for the account you need to
change it right away!
register=a6409sAHZ7088wSG:[EMAIL PROTECTED]
; NOTE: The line below ([gafachi]) can not be changed,
You wil get that error if you hit a system thread limit or run out of
memory. We generally recommend using a 64 bit cpu and os and the
settings detailed:
http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations
Also, not that the default config has sessions per second limited
http://jira.freeswitch.org/browse/MODENDP-61
On Jul 22, 2008, at 7:17 AM, Ivan C Myrvold wrote:
I have a NAT problem with my FreeSWITCH which shows up only when my
public address have changed AFTER I have started FreeSWITCH.
I don't know if this is a small bug in FreeSWITCH, therefore I want
There should be no performance issues with -nf. This is the exact
reason it exists.
Mike
On Jul 23, 2008, at 9:43 PM, John Skopis (Lists) [EMAIL PROTECTED]
wrote:
Birgit Arkesteijn wrote:
Hi all,
We've got an older version of FreeSWITCH (Trunk 7948) running on a
Linux
x86_64
They just ran into the issues this afternoon, I don't think we have a
report into snom yet
Mike
On Jul 23, 2008, at 10:02 PM, Brian Snipes [EMAIL PROTECTED] wrote:
Do you know if any of the snom firmware works as it should for that
functionality or has snom given any kind of estimate on
It seems I have bad ram in my macbook pro and its causing repeated
kernel panics, if anyone is willing to donate to the cause, please
contact me off list.
MIke
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Configure the spidermonkey sub-modules in spidermonkey.conf.xml not
modules.conf.xml
Mike
On Jul 25, 2008, at 7:19 PM, Erol Akarsu [EMAIL PROTECTED] wrote:
I compiled the latest release with --enable-core-odbc-support and
chnaged modules.conf to compile all mod_spidermonkey* modules.
It is an mp3 player that can be embedded in a webpage. If I recall we
use it for the web interfaces to voicemail.
Mike
On Jul 27, 2008, at 11:50 AM, UV [EMAIL PROTECTED] wrote:
The XML-RPC actually works even with the htdocs directory empty.
I’m specifically asking about the slim.swf (and
I have generally just used json/rest type calls, otherwise you would
have to use a lua addin or some code in js. We do have exposed the
curl library so all of the http socket handling should be done, just
no specifics of whatever protocol you want.
Mike
On Jul 28, 2008, at 11:57 AM, Brian
If anyone is a contributor to a source file but their name is not in
the list of contributors for that file, please contact me off list so
that we can get that corrected.
Mike
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to examples?
Where has you used json/rest type calls?
Thanks
- Original Message
From: Michael Jerris [EMAIL PROTECTED]
To: freeswitch-users@lists.freeswitch.org
Sent: Monday, July 28, 2008 12:00:42 PM
Subject: Re: [Freeswitch-users] Calling web services, JMS or EJB
from freeswitch
I
Why do you have a leading / before sofia?
Mike
On Jul 29, 2008, at 11:30 PM, Klaus Teller [EMAIL PROTECTED]
wrote:
Hi,
i have a javscript script that was working with Freeswitch 1.0. The
only issue i had is that outgoing dialing via Gafachi was not
transmitting the caller ID. Then I
again.
klaus.
Original-Nachricht
Datum: Tue, 29 Jul 2008 23:40:14 -0400
Von: Michael Jerris [EMAIL PROTECTED]
An: freeswitch-users@lists.freeswitch.org
freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Gafachi Again
Why do you have a leading
you can set the variables on the originate line with {} , see
http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation
second example.
Mike
On Jul 30, 2008, at 1:24 AM, Klaus Teller wrote:
Thanks Dave,
This however produces following error message:
=
mod_cdr is not a supported module and will not work with the current
api. You can take a look at mod_xml_cdr and mod_cdr_csv which are
supported and in tree,
Mike
On Jul 30, 2008, at 10:14 AM, Shehzad Pankhawala wrote:
Hi every body,
I have downloaded mod_cdr from svn as shown in
On Jul 30, 2008, at 8:33 AM, Sangwoo Jin wrote:
Hi,
I I'm testing freeswitch with sipp.
My test configuration is the following:
Sipp(caller) - freeswitch - Sipp(callee)s
Testing loads are 5 CPS ~ 30 CPS and caller has hanged up a call as
soon as
receiving 200 OK.
In this testing
In comparison to audio handling, the load of the sip connections is
minimal and not worth considering. If reliability is one of your
metrics I would say TCP is more reliable than UDP.
Mike
On Jul 30, 2008, at 12:39 PM, Brian West wrote:
I honestly don't think it'll gain you much if any
-
From: [EMAIL PROTECTED] [mailto:freeswitch-
[EMAIL PROTECTED] On Behalf Of Michael Jerris
Sent: Wednesday, July 30, 2008 11:26 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Memory Leaks
On Jul 30, 2008, at 8:33 AM, Sangwoo Jin wrote:
Hi,
I I'm testing
In the sip_via_host that is going to be the first (next hop) host in
the via. We don't currently go through the entire list of via headers
and turn them into variables. It could be added, but I would want a
compelling reason to add the overhead.
Mike
On Jul 31, 2008, at 4:01 AM, Alois
You are using a recent fedora that ships with a libcurl with broken
dependencies. You either need to install a non broken libcurl (I
don't think they have any non broken packages) or configure freeswitch
with the --without-libcurl option to force freeswitch to not use the
system libcurl.
What advantages does this mod have over just using event socket?
Mike
On Aug 1, 2008, at 10:12, Erol Akarsu [EMAIL PROTECTED] wrote:
Jonas,
This is what I want.
Do you have any documentation on mod_ivr_socket and how its API
looks like?
Thanks
Erol Akarsu
- Original Message
There is also a param in the sip profile:
param name=max-proceeding value=1000/
This is the number of calls in the proceeding state in sip.
Mike
On Aug 1, 2008, at 11:24 AM, Ron wrote:
I have:
param name=max-sessions value=4000/
param name=sessions-per-second value=200/
- Original
I think focusing on just the open source projects may be missing the
mark. I would be much more interested in a more detailed feature,
performance, and other metrics comparison between open source and the
commercial products in the same market segments (ip PBX, call center,
sbc and soft
Yes and no. It's an entirely new invite where some information is
passed from one side to the other, not modifying the invite and
sending it back out, a subtile but important difference.
Mike
On Aug 3, 2008, at 2:07 PM, Robert Dyck wrote:
I read that FS is a B2BUA and not a proxy. However
freeswitch already has a system dialplan application and a strftime
fsapi command so you should be able to directly reproduce exactly what
you have below using dialplan. I might also look at adding a curl
dialplan application and/or fsapi command as those could be pretty
useful.
Mike
I don't think any of the bugs that were posted were posted as security
risks, but with hundreds of bugs fixed, I am sure that some were
remotely exploitable. I will try to do a better job marking them in
changelog in future versions.
Mike
On Aug 3, 2008, at 2:37 PM, John Skopis (Lists)
I have used ixia gear in the past to do this sort of testing but it's
not cheap. Do you have access to any commercial test gear?
Mike
On Aug 8, 2008, at 7:38 PM, Brian West [EMAIL PROTECTED] wrote:
I think this one has rtpecho on. You really can't do pcap replay sipp
crashes before you
It should already work with what is in opezap using boost but we have
not done the testing on it yet. If someone has cards and a line I
would love to hear results.
Mike
On Aug 6, 2008, at 1:24 PM, Michal Bielicki [EMAIL PROTECTED]
wrote:
It would or it will ?
On Aug 6, 2008, at 8:14
He is saying to add the var to the user in the variables so when we
auth that user, the variable is set.
Mike
On Aug 6, 2008, at 11:58 AM, UV [EMAIL PROTECTED] wrote:
I have to agree with Roberto on not understanding what you’ve done,
Brian.
Is toll_allow an undocumented channel
The java api is 99% the same as the python/lua/perl api. You can
check the wiki and the sample scripts in the source tree. This may be
incomplete, feel free to ask any questions here where you can't find
samples and we can try to fill in the missing pieces if they are not
on the wiki.
Pekka Pessi from the sofia-sip project pushed a patch that is now
synced to our tree. Please update and give it a try.
Mike
On Aug 11, 2008, at 11:41 AM, Michael Jerris wrote:
Try posting this to the sofia-sip mailing list, lets see if we can
sort out a fix there.
Mike
On Aug 11, 2008
When you run any of the embedded languages like this as an app we
magically create an object called session for you of the session
that is running the app. Do the same thing you did in your script but
don't create csession just use session instead.
Mike
On Aug 17, 2008, at 5:34 AM, Sheeju
All of this is already done for you in the switch, both the
decrementing of max forwards and the rejection as well. You can check
out code in switch_core_session_outgoing_channel and
switch_ivr_session_transfer for more details.
Mike
On Aug 19, 2008, at 9:38 AM, Jonas Gauffin wrote:
Hi
Try with svn revision 9321 or later.
Mike
On Aug 19, 2008, at 10:41 AM, kokoska rokoska wrote:
Brian West napsal(a):
That would indicate your curl wasn't compiled with SSL support
eh? ;)
I don't thik so, because (like I wrote before) mod_xml_curl works fine
with SSL on the same
I have responded before and I will respond again. No one has ever
added TLS support to the build. It requires adding openssl into the
windows build process. Feel free to add any patches enabling this to
a jira.freeswitch.org bug and I would be happy to take a look.
Mike
On Aug 20, 2008,
We don't keep a comparison but we do have our full list:
http://wiki.freeswitch.org/wiki/Mod_conference
Mike
On Aug 20, 2008, at 4:48 AM, Gavin Henry wrote:
Hi all,
Is there a comparison of each or anythign obvious?
We've got freeswitch install here for testing, just getting our
heads
Typically no audio is related to some sort of nat problem. Can you
get a full debug log and sip trace of the call and pop by on
irc.freenode.net (#freeswitch) and someone should be able to help.
Mike
On Aug 20, 2008, at 8:26 AM, msp wrote:
Hi all,
I need to make calls using bgapi
We have the fsapi interface which is available via the console, event
socket, xml-rpc, http and others. It is not exposed to sip itself.
Can you describe a bit of what your use case is for this?
Mike
On Aug 20, 2008, at 4:23 AM, Gayatri Kulkarni wrote:
Hi guys,
Is there a way
On Aug 20, 2008, at 11:53 AM, Gavin Henry wrote:
2008/8/20 Anthony Minessale [EMAIL PROTECTED]:
One page you can look at is this one:
http://www.freeswitch.org/node/100
This is a comparison matrix of all of the existing Asterisk based
conference
apps including app_confcall which is my
I'll remove you manually.
Mike
On Aug 20, 2008, at 2:11 PM, Chris Williams wrote:
How do I get off this mailing list? I am not a programmer and have
no idea why I’m signed up for this mailing list. If someone can
help, please email me directly and let me know what to do.
Thank you.
On Aug 20, 2008, at 2:23 PM, Ken Livingston wrote:
Ironically, it was not in the message sent by Chris.
Yes it was.___
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We are already auto-adjusting to the nat issue here, but you likely
have a firewall on blocking the rtp traffic, please turn off your
firewall and test again.
Mike
On Aug 21, 2008, at 4:02 AM, Adeel Ansari wrote:
I am not using any ip phone. Its normal mobile phones.
Moreover, I tried to
The proper prototype is file_name time_limit silence_threshold
silence_hits
recordFile(char *file_name, int time_limit = 0, int silence_threshold
= 0, int silence_hits = 0);
On Aug 21, 2008, at 1:19 AM, Marc Orenberg wrote:
I sent this email this morning but it didn't seem to make it
wanted to know is there anyway by which FS publishes
it's functionality to other network elements?
I suppose the fsapi interface you have mentioned should be a way of
doing it but not sufficient - correct?
On Wed, Aug 20, 2008 at 8:40 AM, Michael Jerris [EMAIL PROTECTED]
wrote:
We have
Are you talking about speech recognition? Utterance or dictation?
On Aug 21, 2008, at 2:56 AM, Ilan Perez wrote:
Is anyone working on doing this?
Ilan Perez
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