What do the debug logs on fs say when you try to put the call on hold?
Mike
On Aug 6, 2009, at 1:01 PM, Kozak Vladimir wrote:
The scenario is the following:
FS User A dial an extension
Extention opens outbound socket channel to my application
My application bridges the call to FS User B
The
typically when these questions are asked we find people really want to
be doing it the way these signals are automatically passed across.
Can you describe what your trying to do a bit more?
Mike
On Aug 7, 2009, at 7:44 PM, Max Bridgewater wrote:
Hi,
using javascript, i do originate the
I think that summary is totally wrong. Loopback should be used
here, and this should work to do what you want, just be aware of what
that means.
Mike
On Aug 8, 2009, at 4:24 PM, Phillip Jones wrote:
Mike/Rupa ,
Thanks for your help on this. So I am correct that summarizing that
please open a bug on jira.freeswitch.org with the details of exactly
how to re-create this issue
Mike
On Aug 8, 2009, at 5:21 PM, Benedikt Fraunhofer wrote:
Hello Mike,
2009/8/8 Michael Jerris m...@jerris.com:
how many does it stop at? is it the same number each time?
i tried
http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP
turn the logging all the way up and see what it says.
Mike
On Aug 4, 2009, at 2:51 PM, Matthew Fong wrote:
Hi Mathieu, thanks for the reply. I enabled sip trace logging and
got the logs below, but I am still at a loss at being able
Please report this error to centos bug tracker, aspell-devel should be
a dependency of php-devel.
Mike
On Aug 5, 2009, at 10:40 AM, Greg Thoen wrote:
Trying to make phpmod and it fails with this:
/usr/bin/ld: cannot find -laspell
collect2: ld returned 1 exit status
make[1]: *** [ESL.so]
you could probably pull off the same thing with xml_curl for dialplan
and a simple set of bridge and then transfer in the actions.
Mike
On Aug 5, 2009, at 11:54 PM, Woody Dickson wrote:
Hi,
In my module, I will collect a list of available failover route that
I can use to failover to
What is the output of wantouter hwprobe?
On Aug 3, 2009, at 3:12 AM, Merul Patel me...@mac.com wrote:
I'm new to FS, and experimenting with it on a constrained
environment (PCEngines ALIX board running Voyage Linux 0.62).
So far, FS has compiled fine, and I can register multiple
softphones
It is still there.
On Aug 3, 2009, at 8:38 AM, afshin afzali a.afzali2...@gmail.com
wrote:
Hi,
I'll appreciate if somebody tell me where has gone the mod_curl ? I
just need to use it for http method calls.
Regards,
-- afshin
___
I have yet to configure one of these cards for freeswitch so it's
possible it's not in the config util yet. I suggest contacting
sangoma support for assistance
On Aug 3, 2009, at 12:56 PM, Merul Patel me...@mac.com wrote:
What is the output of wantouter hwprobe?
voyage:~# wanrouter
On Aug 2, 2009, at 5:34 PM, Merul Patel wrote:
I'm new to FS, and experimenting with it on a constrained
environment (PCEngines ALIX board running Voyage Linux 0.62).
So far, FS has compiled fine, and I can register multiple softphones
and make calls between them, but I'm lost at how to
we don;t have any german sound files?
On Jul 31, 2009, at 10:31 AM, Rudolf Denert wrote:
Hi again,
does anybody know why my freeswitch doesn't play German speech-files?
Here is my construct:
I'm generating a random number in my lua script:
rand = math.random(11, 1000);
The I want that
are you answering the call somewhere or running any other dialplan
actions that may establish media before you set proxy?
MIke
On Jul 30, 2009, at 11:05 AM, Stefano Marinelli wrote:
Mathieu Rene ha scritto:
You need
param name=inbound-late-negotiation value=true/
Yes, I've tried both
The far end is sending a bye for no clear reason I can see in this log.
Mike
On Jul 30, 2009, at 10:15 AM, Stefano Marinelli wrote:
Hi.
I'm trying to receive a fax using Freeswitch. It's a SIP channel. I
know
there's no T38 support (yet) but I think I can get decent result with
G711 (as
In the header files?
On Jul 29, 2009, at 11:11 PM, Thangappan.M thangappan...@gmail.com
wrote:
Have you got my questions.?
Using ESL connection object some of the function or subroutines or
methods has bee called. Is there any way to find out all the
function names and its
You must turn on the option to manage presence in the sip profile.
Mike
On Jul 28, 2009, at 5:43 AM, Gregory Charles
gregory.char...@sogeti.com wrote:
Hi everybody,
I intend to use Freeswitch with two Ekiga Softphones. SIP Instant
messaging works between the two softphones but SIP
using 95 is wrong. That is not part of the dynamic range for
unassigned codecs. This needs to be fixed on their side.
MIke
On Jul 28, 2009, at 12:23 PM, Brian West wrote:
On Jul 28, 2009, at 11:14 AM, Helmut Kuper wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
today I
also take a look at execute_on_answer if you want it to be scheduled
from answer instead of from that point in the dialplan.
Mike
On Jul 28, 2009, at 1:48 PM, Saeed Ahmad wrote:
action application=sched_hangup data=+600/
On Tue, Jul 28, 2009 at 7:28 PM, Kristian Kielhofner
Did you read his response to you? Please generate a usable backtrace
as rupa explained and post a bug to jira freeswitch.org
On Jul 27, 2009, at 9:00 AM, Baskar yudha2...@gmail.com wrote:
Hi Rupa,
I get core dump segmentation fault in freeswitch machine
frequently. can any one assist
mod_iax isn't loaded. I suggest using sip anyways.
Mike
On Jul 27, 2009, at 1:23 AM, velusamy velu wrote:
Dear All,
I have tried to call a Asterisk extension from FreeSWITCH. I
have done the following configurations,
* I have enabled mod_iax module in
On Jul 24, 2009, at 11:45 AM, Gu Sh wrote:
I have been using freeswitch for over a year and I love all of the
features, extensibility etc. Recently one of the clients wanted to
use a IAX client and call from the IAX client works fine but there
was one feature requested by my client
It is not built by default because it requires manual intervention to
make sure you have a proper threadsafe perl and all its dev libs
installed first. We work hard to make sure all default modules build
out of the box with minimal external dependencies. Also, this module
still does not
because your not running limit at all when you are doing an originate
directly. You can use loopback to originate through a dialplan
extension.
Mike
On Jul 22, 2009, at 8:45 AM, Eli Hayun wrote:
Hi
I set the limit to 1 on the extension like that
action application=limit_hash
Do you have anything on that extension?
On Jul 22, 2009, at 7:21 PM, Luis F Urrea lfur...@gmail.com wrote:
I don't know if this may be related but in voicemail.conf.xml by
default the two params that follow are defined:
param name=operator-extension value=operator XML default/
param
This is not correct. Do you have the odbc mod for spidermonkey loaded?
On Jul 20, 2009, at 9:36 AM, Meftah Tayeb tayeb.mef...@gmail.com
wrote:
hello baskar,
i think Freeswitch ODBC Support is not enabled for Windows
you must compile it with ODBC Support enabled
thanks
Baskar wrote:
*Hi
See the chat_send api command and it also should just work with
presence peers.
On Jul 19, 2009, at 3:26 AM, Woody Dickson woodydick...@gmail.com
wrote:
Hi,
I would like to use freeswitch as a gateway for sending and
receiving short message.
Does Freeswitch have the capability to
Please try looking on the wiki, this and many other questions should
be answered for you there.
Mike
On Jul 15, 2009, at 4:05 AM, Brad Tuan wrote:
How to set the date format , and the IVR flow ??
___
Freeswitch-users mailing list
We have no qsig support.
On Jul 12, 2009, at 9:54 AM, Maarten De Maeyer i...@nalawo.com wrote:
Hi,
Can someone tell me how complete QSIG support is in FS ? Is there a
config example available ? I need to connect FS to another pbx with
QSIG.
Any tips/advice more than welcome.
Thanks.
See mod_xml_curl
On Jul 12, 2009, at 4:45 PM, Edward Q. q.edw...@gmail.com wrote:
Hi guys.
I was just wondering if it is possible to have the
outbound_caller_id dynamically pulled from MySQL db ?
If it is can anyone please point me in the right direction ?
Thanks in advanced to all for
could you post how you tired to do it in dialplan that didn't work?
Mike
On Jul 10, 2009, at 5:57 AM, Mark Campbell-Smith wrote:
Hi!
1. Can I email the voicemail message to multiple email addresses?
A comma separated list does not work in the extensions.xml file
(1000.xml), but it does work
Look closer at the logs, we don't send a 200ok in a bridge until we
get one from the b leg.
Mike
On Jul 10, 2009, at 5:39 AM, Kozak Vladimir wrote:
Hello,
I have the following problem: I send Invite without SDP to
Freeswitch on destination_number xxx_123
extension name=super_test
mute/unmute is a toggle.
Mike
On Jul 9, 2009, at 11:20 AM, Michael Collins wrote:
2009/7/9 Кривушин Михаил krivushi...@rn-
inform.tomsk.ru
Hello!
Is any ability to ask to unmute in conference?
Not sure if I understand the question. Are you talking about the
caller pressing zero to
Looking closer, it looks like it ran the dialplan, executed some
actions to set vars and ran out of actions to do so it hung up.
On Jul 3, 2009, at 1:25 AM, Edmar Cruz darklio...@yahoo.com wrote:
The same issue...
param name=enable-3pcc value=true/
Michael Jerris wrote:
Try enabling
I would guess your looking to do more than set variables, you should
make it do those other actions in the dialplan
On Jul 3, 2009, at 2:32 AM, Edmar Cruz darklio...@yahoo.com wrote:
What do you think I shall do?
Michael Jerris wrote:
Looking closer, it looks like it ran the dialplan
It can go to anything that works with odbc theoretically.
On Jul 3, 2009, at 6:30 AM, Edmar Cruz darklio...@yahoo.com wrote:
Is there any options make nibble rates to MySQL database?
--
View this message in context:
http://www.nabble.com/Database-for-Nibble-Rates--tp24320955p24320955.html
The log you had in this thread never called bridge at all.
On Jul 3, 2009, at 11:30 PM, Edmar Cruz darklio...@yahoo.com wrote:
I think the problem is on the bridge
action application=bridge value=sofia/default/$...@116.454.20.12/
Am not dialing a registered user so I put $ instead of %
On Jul 2, 2009, at 7:50 PM, Steve Underwood wrote:
If by the usual way you mean the standard 2 + 2 letter codes we are
used to on computers, that just doesn't work. As I said before, those
are for written languages, not spoken languages. There are no standard
codes for many spoken
Try enabling 3pcc in the sip profile.
On Jul 2, 2009, at 11:12 PM, Edmar Cruz darklio...@yahoo.com wrote:
I have a GSM gateway. The issue is sometimes the calls failed what
is the
cause of the error this is my logs?
This is on my freeswitch logs...
09-06-25 10:21:50 [DEBUG]
There was a bit of work towards it but no one has worked on it lately
On Jul 1, 2009, at 4:24 AM, François Delawarde fdelawa...@wirelessmundi.co
m wrote:
Is there any work planned for T.38 termination (in mod_fax)?
François.
On Tue, 2009-06-30 at 12:07 -0400, Michael Jerris wrote:
We
How much memory is it using? Can you use memstat to see where the
memory is allocated.
Mike
On Jul 1, 2009, at 8:29 AM, Muhammad Danish Moosa
danishmo...@gmail.com wrote:
Hi
Freeswitch is being used in a scenario where two endpoints are
running traffic with bypass media mode.
I think that detection is not working on mac right due to it looking
in default search paths. I am in process of fixing this to use in
tree libtiff soon so this should fix this issue.
Mike
On Jun 29, 2009, at 11:08 PM, seven wrote:
Hi,
I'm on the latest svn 14041, and I have tiff
the bridge app already does all this for you doesn't it (along with
bind_meta) ?
Mike
On Jun 27, 2009, at 2:45 AM, Dome Charoenyost wrote:
Dear All,
I try
s = new Session(sofia/external/x...@xxx.xxx.xxx.xxx);
if (s.ready()){
s.setVariable(nibble_rate, 2.5);
If you don't need authentication, you don't need a gateway, if you do,
you will need to setup a user on the other box to register to.
On Jun 30, 2009, at 3:05 AM, Brad Tuan wrote:
I know that i need to set the dialplan,
my problem is when FS_B send a REGISTER to FS_A, FS_A will return a
http://wiki.freeswitch.org/wiki/Mod_commands#in_group
http://wiki.freeswitch.org/wiki/Mod_commands#user_exists
On Jun 30, 2009, at 6:09 AM, Christian Benke wrote:
Hello!
I have the following scenario:
I want to check if a called extension is part of a group, or as an
alternative, if it is
you have a pointer somewhere in your directory for that user, hard to
see without seeing the whole config, but grep for 111 and see what
else you find.
Mike
On Jun 30, 2009, at 10:21 AM, Alexey Lubimov wrote:
I sofia_reg.c:1765 have two user records - good #110 and bad #111.
bad.xml:
We currently support t.38 passthrough only using proxy_media mode. T.
38 gateway is on the roadmap but not yet close to complete.
Mike
On Jun 30, 2009, at 5:15 AM, François Delawarde wrote:
Many issues on Asterisk's T.38 (or probably just on T.38?)...
Could it convince those relying on
If you are interested, we can create a contrib directory for you in
the codebase and you can commit it right in tree. If interested,
please log on to irc and we can coordinate setting up an account for
you.
Mike
On Jun 29, 2009, at 3:23 AM, Prabhuram Mohan wrote:
Hi All,
Abode AIR
The application interface doesn't return a status like that.
Different applications may set channel variables on an app by app
basis. if your doing a bridge you can look at for example:
http://wiki.freeswitch.org/wiki/Channel_Variables#originate_disposition
Mike
On Jun 27, 2009, at 7:47
I would be curious if you have debug output of a call that is
returning NONE there. That seems like we should be setting a hang-up
cause somewhere.
Mike
On Jun 28, 2009, at 12:16 PM, Nicolas Brenner wrote:
I have a small JS script that makes a call, plays a sound file and
then hangs up.
Fifo is not configured through javascript, it is configured via the
configuration files.
http://wiki.freeswitch.org/wiki/Mod_fifo
Mike
On Jun 29, 2009, at 12:53 AM, Baskar wrote:
Hi,
I have configured inbound through JavaScript it is working well.
Through dialplan i have configured FIFO
Posting a quick sip trace of the failed register will probably be
helpful. Also, are the debug logs any help?
Mike
On Jun 29, 2009, at 11:11 AM, Frederik Denkens wrote:
Hi,
Following the recommendations of this list, we went with an external
gateway to connect to the BRI based ISDN
If you come up with a good configuration, we would appretiate it if
someone could post this to the wiki.
Mike
On Jun 29, 2009, at 11:52 AM, Christian Löschenkohl wrote:
hi
i'm inalp patton certified, so maybe i can help
if you can post your config (export startup config).
please also
I don't think we expand vars here, you will need to expand us-ring
yourself and pass the string
Mike
On Jun 29, 2009, at 5:56 PM, Nik Middleton wrote:
One little annoyance though, I cannot for the life of me get a
ringback tone while the B leg is ringing,
I’ve tried putting
-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Nibblebill and multiple gateway
2009/6/26 Michael Jerris m...@jerris.com:
I said to just add the set import=nibble_rate, your re-setting it
for
no reason (and getting rid of the change that should have helped) by
your import
most of the information about fifo is : http://wiki.freeswitch.org/wiki/Mod_fifo
Mike
On Jun 26, 2009, at 2:45 AM, Dome Charoenyost wrote:
Dear All,
I'm asterisk developer(I have some code in Asterisk) .After 3
weeks with freeswich nothing to say. now i'm move all callingcard ,
On Jun 26, 2009, at 10:17 PM, Vincent Stemen vince.freeswi...@hightek.org
wrote:
On Wed, Jun 24, 2009 at 12:53:23AM -0400, Michael Jerris wrote:
Can you post a bug to Jira.freeswitch.org with all these warnings,
even better with patches to fix it.
OK. I think I have narrowed
=+dialprovider_id[1]);
bridge(session,s);
}
and check hangup cause before try other provider.
Please guide me it's right way or not ?
Dome C.
2009/6/26 Darren Schreiber d...@d-man.org
Did this work? Would love an update on this error/issue.
From: Michael Jerris [mailto:m...@jerris.com]
Sent
On Jun 25, 2009, at 5:49 PM, Vincent wrote:
On Tue, Jun 23, 2009 at 11:19:51PM -0400, Andrew Thompson wrote:
On Tue, Jun 23, 2009 at 09:15:30PM -0500, Vincent Stemen wrote:
Ok. I did this.
Compilation still failed but there are significant improvements
since
the last time.
Here is
Of course not. This is why many do billing in icrements like
mod_nibblebill does. Radius (although not yet with our module) and
diamater both work this way and solve this issue. This in combination
with session timers adress this and the hangup issue during a
catastophic switch or
try adding
action application=set data=import=nibble_rate/
before the bridge and report back results.
Mike
On Jun 24, 2009, at 1:36 AM, Dome Charoenyost wrote:
Dear All,
Look like nibblebill does't work with multiple gatreway.
I try
action application=set
If you can supply a patch to expose this as a config option for us it
would be appreciated. Patches can be posted to http://jira.freeswitch.org
.
Mike
On Jun 17, 2009, at 3:22 PM, Muhammad Shahzad wrote:
Ok, thanks, i will take care of it in my code where necessary.
Thank you.
On Thu,
Try turning up your logging level to debug to see why the call is
hanging up.
Mike
On Jun 19, 2009, at 7:13 AM, Edmar Cruz wrote:
My freeswitch has a mysql database consists of freeswitch tables,
registrations and nibblebill on mysql configured it correctly and
working...
Issue is when
if you need to use the same tags, we should be using the whole same nh
in the code. There is code to do this by call uuid but I can't recall
if thats for NOTIFY or INFO. If its the wrong one, we should add teh
same for what you need.
Mike
On Jun 21, 2009, at 6:05 AM, Christian
if you turn up the debug logs it should tell you why.
On Jun 22, 2009, at 11:38 PM, Edmar Cruz wrote:
Nope. I just want to call a mobile number with no register number.
Brian West-3 wrote:
I'm going to guess you're calling a registered user? If so replace
the @ with %
/b
On Jun 22,
Please see the debugging pages on the wiki
On Jun 23, 2009, at 10:10 PM, Edmar Cruz darklio...@yahoo.com wrote:
Where can i find this logs?
Michael Jerris wrote:
Try turning up your logging level to debug to see why the call is
hanging up.
Mike
On Jun 19, 2009, at 7:13 AM, Edmar Cruz
On Jun 23, 2009, at 10:15 PM, Vincent Stemen vince.freeswi...@hightek.org
wrote:
Thanks for the response Anthony.
On Tue, Jun 23, 2009 at 08:29:13AM -0500, Anthony Minessale wrote:
You are way off base in a few places, let me see if I can clarify a
bit.
Here are at least 2 pointers:
It has both the interface bind address and the external address
Mike
On Jun 21, 2009, at 3:38 AM, Nandy Dagondon g...@i.ph wrote:
it's working now, i mean the Auto NAT feature - after i enabled UPNP
feature on my router. it's based on external IP addresses of Ext-SIP-
IP and Ext-RTP-IP when
It's failing to build the core library. There should be some warning
before it tried to build the modules in the log.
On Jun 21, 2009, at 9:25 PM, Joseph L. Casale jcas...@activenetwerx.com
wrote:
I attempted to build rpm's from the included spec file using a non-
root user
build
Some of these things make sense is some scenarios but not others.
Most people are wanting to do full topology hiding, so we don't by
default pass very much across a bridge. I am interested in working on
this, feel free to contact me off list with your findings.
Mike
On Jun 17, 2009, at
Try using loopback endpoint for this test .
Mike
On Jun 17, 2009, at 10:00 AM, Tim B wrote:
Trying to do a local test for faxing. Keep getting an error.
default dialplan:
extension name=test_rxfax_stream
condition field=destination_number expression=^8000$
action application=answer /
action
If you can catch brian or me on irc can you provide remote access to
this box and we should be able to fix this pretty quick
Mike
On Jun 16, 2009, at 5:20 AM, seven dujinf...@gmail.com wrote:
Hi brain,
Are you still looking into this?
I think it must be some error when it register, I
The only way I can think to do this today would be to cancel the call
and re send with the intercom headers for a phone that supports it.
It may be possible to send a reinvite with autoanswer headers but I
doubt that would work, all you could do is try making code to do it it
a sipp or
.
But what if I
* transfer it to the same user destination again (now with intercom
enabled), will this work?
* transfer it to park and then transfer it to the same destination
again (now with intercom enabled)
Best regards
Peter
Michael Jerris schrieb:
The only way I
This issue is now fixed in svn. Thanks Seven for access to your box
to troubleshoot.
Mike
On Jun 16, 2009, at 9:17 AM, Seven Du wrote:
What's wrong of the contact string? 639(snom) works but 637(zoiper)
doesn't.
user
-Authenticate: Digest realm=sip2.mycompany.de,
nonce=2ee26efe6ab27f88, algorithm=MD5
Content-Length: 0
and hangs up.
Anybody know how to solve this Snom intercom issue?
Best regards
Peter
Michael Jerris schrieb:
The transfer should work but it sounds like offhook agents is what
your
http://wiki.freeswitch.org/wiki/Freeswitch_XML-RPC
http://wiki.freeswitch.org/wiki/Mod_commands
http://wiki.freeswitch.org/wiki/Mod_conference
Mike
On Jun 12, 2009, at 3:35 AM, Santhosh wrote:
Hi,
Is there any where I can find more documentation on the XML-RPC
interface of freeswitch. I am
On Jun 11, 2009, at 2:24 PM, John Dalgliesh wrote:
On Thu, 11 Jun 2009 at 10:42 -0700, Michael Collins wrote:
On Thu, Jun 11, 2009 at 10:33 AM, Michael Giagnocavo m...@giagnocavo.net
wrote:
Exactly. You probably want to have something like this anyways, so
that
when someone
If they were still showing in status, can you use gcore to dump a core
next time this happens, leave it running somewhere we can get to it
and post a thread apply all bt to Jira.
Mike
On Jun 11, 2009, at 5:40 PM, Nik Middleton nik.middle...@noblesolutions.co.uk
wrote:
It was the output
the default alias was removed from the default configs last week, so
new configs don't have this anymore.
On Jun 10, 2009, at 10:54 AM, Max Bridgewater wrote:
Thanks,
the first variant doesn't work for me. Any idea?
I changed it to:
originate sofia/internal/1...@192.168.10.103 park()
http://wiki.freeswitch.org/wiki/Dialplan_XML
break=on-true ?
On Jun 10, 2009, at 12:18 PM, Max Bridgewater wrote:
Thanks Folks; I'm making progress. The following origination string
does make my non-registered SJPhone ring:
your svn update failed,
rm -rf libs/pcre svn update ./bootstrap.sh ./configure
make current
On Jun 10, 2009, at 2:30 PM, Nik Middleton wrote:
Hi Guys,
Ran make current today, and am getting the following errors. I ran
bootstrap and configure, but still get these messages.
Any
On Jun 5, 2009, at 4:10 PM, Shoaib Khanzada wrote:
Hi,
How can I use setGlobalVar and getGlobalVar in my javascript to
store a ODBC connection?
I want to set an ODBC database connection object globally so that I
can access it from anywhere. This connection will be used for read-
only so
java routing script?
You might be able to send a 100 trying from your dialplan or script.
Keep in mind that this message stops timers on the originating side.
- original message -
Subject:Re: [Freeswitch-users] Freeswitch creating more then two
sessions for one call
From:Michael
http://wiki.freeswitch.org/wiki/Reporting_Bugs
Please attempt to reproduce this issue with trunk with crash
protection disabled, and if you are able please file a jira with a
backtrace of the crash
Mike
On Jun 4, 2009, at 4:23 AM, Andy Ayers wrote:
Hi,
Every few days I'm getting this
Can you please re-test with current svn trunk. we added some new nat
busting code yesterday that may assist with this. You will need to
specify the new param name=local-network-acl value=localnet.auto/
param in the sofia profile (see
Are you having this issue on your analog or pri lines? what does your
openzap.conf look like?
Mike
On Jun 4, 2009, at 4:28 AM, god.nirvana wrote:
hi all
i am new to freeswitch.
there are some busy tone detect issues,i hope someone could help me.
i installed freeswitch from
if you could nail down a specific svn revision that causes this issue
and file a jira at http://jira.freeswitch.org that would be a big help
in resolving this issue.
Mike
On Jun 3, 2009, at 3:43 PM, Bruce McAlister wrote:
Hi All,
Any pointers or suggestions on this issue would be greatly
can you try in the square brackets using http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_number
instead? I think effective will work if you set it but not in the
square brackets.
Mike
On Jun 2, 2009, at 7:04 AM, Yuriy Ivzhenko wrote:
Hello all.
I have test the lcr
/usr/local/freeswitch/bin/fs_cli -x reloadxml
On Jun 2, 2009, at 4:04 AM, Brad Tuan wrote:
How to reload xml without using console command line??
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
apiExecute is for api commands (the ones you can run at the cli) and
session.execute is for applications (the ones you run from the
dialplan).
Mike
On May 31, 2009, at 1:43 AM, jcro...@gmail.com wrote:
From JavaScript, what is the difference between apiExecute and
session.execute?
It
Can you try to do a binary search and nail down the exact version that
caused this issue and then file a bug on http://jira.freeswitch.org.
Thanks
Mike
On May 29, 2009, at 9:55 AM, Peter Olsson wrote:
I’m on Windows, so I have everything under my fs directory, but I
deleted the complete
apr is used primarily for portability and utility, not performance.
It is used heavily throughout the entire codebase in the core and all
modules (via our abstraction layer).
On May 28, 2009, at 10:14 AM, Gopalakrishnan A.N wrote:
Hi,
I saw the apache portable runtime is included in
This looks like a permissions error on creating or opening files in
the db directory.
Mike
On May 25, 2009, at 12:25 AM, mashudi wrote:
Hi Guys,
I have install Freeswitch with version : FreeSWITCH Version 1.0.4pre7
(13238M) Started.
I load the openzap module after install the wanpipe
try the 3.5.2 drivers.
Mike
On May 15, 2009, at 12:33 PM, Brian Wood wrote:
I have a Sangoma A200 with 1 FXO and 1 FXS module. Previously, I was
using zaptel under the wanpipe-3.2.7 drivers. It worked fine in this
configuration, but DTMF recognition was a bit flakey.
I am trying to switch
zap_module = {
wanpipe,
wanpipe_init,
wanpipe_destroy,
Michael Jerris wrote:
try the 3.5.2 drivers.
Mike
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do you see any errors if you do a make ?
Mike
On May 13, 2009, at 8:56 AM, Saeed Ahmed wrote:
Hi Tristan,
No its not commented out.
And strange thing is that it was working until I did ‘make current’
today.
Thanks for your response.
From: freeswitch-users-boun...@lists.freeswitch.org
the 1.0.9 sounds were rolled tonight and they contain these fixes.
Mike
On May 5, 2009, at 8:38 AM, Peter P GMX wrote:
I looked at my install directory and in the source files
(freeswitch-sounds). No file of this name there.
Thanks for the link. Now it works.
Best regards
Peter
Brian
Do you have any specifics of the errors?
Mike
On May 1, 2009, at 12:07 AM, mszla...@aol.com wrote:
I'm getting Windows Express compiler errors on the latest svn update
to trunk 13213.
It looks like the path is wrong to some files.
Instead of folder Debug, it's looking for files in folder
I can confirm that we comply with rfc 5411 in that we agree that is a
list of sip specs that we may or may not honor, and that we may or may
not have ever seen or read. Joking aside, the sofia list is pretty
good, there are some things noted as it would be implemented in the
application.
Can you point out any place we do sub milli second sleeps? The timer
thread should be doing 1ms, I can't think of any that would be less.
MIke
On Apr 30, 2009, at 2:28 PM, Paweł Pierścionek wrote:
Hi,
With really old kernels (100Hz) if You do sleep(1ms) You sleep for
10ms on average.
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