Hi, Any help or suggestion regarding my previous post. Especially
I also noted that, if I don't receive any events, especially
SERVER_DISCONNECTED, then the connection is in established state, but once
I receive the SERVER_DISCONNECTED event, the connection is closed. Is it
correct??
Here is the
Your using outbound socket and you hangup the call, so it tells you it
is done with the server disconnected message and drops the
connection. This is all as expected. I guess I don't understand what
you think is the problem. This code is doing exactly what I would
expect it to do.
Hi
Checked out svn checkout y'day. I am in the UK. Installed . Installed on
Fedora 11 i386 box.
:
bootstrap.sh
configue --without-libcurl
make
make install
On startup only errors: PMP I'm not behind a NAT so OK
Stacksize registered as too high and advised to use the -waste switch.
Other than
This is resolved. I could someone to call my VOIPUSER number and call
transferred to my designated extension. I could not get this to work
from my network ie calling from one of my extensions and setting that
the call be -rerouted to another extension.
All OK now.
Thanks folks
Otis wrote:
I got a freeswitch that is behind nat and got three profiles.
External (all calls are going through a proxy):
param name=rtp-ip value=$${local_ip_v4}/
param name=sip-ip value=$${local_ip_v4}/
param name=ext-rtp-ip value=$${external_rtp_ip}/
param name=ext-sip-ip
Hi folks
Can I get FS to re-route incoming-calls to PSTN. If this has been
raised before could someone direct me to URL or link please
Thanks.
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Hi Everyone,
How do I get the outbound sofia SIP route that the call took into a CDR
when using fail over dialing with the 'bridge' application?
...I have tried numerous approaches with no luck, this last attempt pasted
below did not work either:
dialString =
Thanks. I will try it . I am on Fedora 11
Mark Crane wrote:
how about trying Fusionpbx.com ( GUI) -Ram
I'll second that! I released FusionPBX 1.0 RC5 today. I thought it was
ready to release now but decided to do one more release candidate just
to be sure. This should be the last
Thank you very much MC . Its working :) . I started loving FS ;)
On Wed, Nov 25, 2009 at 9:25 AM, Michael Collins m...@freeswitch.org wrote:
On Tue, Nov 24, 2009 at 6:03 PM, Lei Tang lei.tl...@gmail.com wrote:
you can do this in follow steps:
1.edit default.xml diaplan config file in your
You need to use brackets [] not braces {} for per-leg variables.
On Thu, Nov 26, 2009 at 6:08 AM, Michael Toop
micha...@voxcore.voxtelecom.co.za wrote:
Hi Everyone,
How do I get the outbound sofia SIP route that the call took into a CDR
when using fail over dialing with the 'bridge'
Fedora and Centos installation instructions are very similar. You should be
able to compile on Fedora without any problems that I'm aware of.
Regards,
Nik
On Thu, Nov 26, 2009 at 06:24, Otis ab...@greatiam.com wrote:
Thanks. I will try it . I am on Fedora 11
Mark Crane wrote:
how
Are you doing this all on a linux box thats acting as your router
too? If not you don't need two profiles... you also don't need to set
the local-network-acl on ANY profile that isn't do anything with nat.
/b
On Nov 26, 2009, at 5:03 AM, Jonas Gauffin wrote:
I got a freeswitch that is
Is there any way to build a dial plan so that when an extension calls
itself the call is automatically put to that users voice mail?
Example, extension 1001 calling 1001 and is sent to voice mail (to
receive messages).
I know that there is a * code to get to voice mail, I cannot recall
which one
It's a windowsserver which is behind a router.
Which profile should local-network-acl be specified on?
When I bridge calls to the outside world, should I use
sofia/internal/phoneNumber@gateway or
sofia/external/phoneNumber@gateway?
On Thu, Nov 26, 2009 at 4:42 PM, Brian West
On Tue, Nov 24, 2009 at 5:48 AM, Eliot Gable
egable+freeswi...@gmail.com wrote:
Or, you can use something like Smarty to cache your generated XML on
your web server and only invalidate those cached results when you
change something that will impact them.
That sounds like a pretty sane way to
In this case you should not need 2 profiles either.
On Nov 26, 2009, at 1:14 PM, Jonas Gauffin wrote:
It's a windowsserver which is behind a router.
Which profile should local-network-acl be specified on?
When I bridge calls to the outside world, should I use
Of course. Please read through the default configs and the getting started
guide and xml dialplan information on the wiki.
Mike
On Nov 26, 2009, at 12:38 PM, Orien Love wrote:
Is there any way to build a dial plan so that when an extension calls
itself the call is automatically put to that
http://dev.mysql.com/doc/refman/5.1/en/connector-odbc-news-3-51-18.html
MySQL Connector/ODBC now supports batched statements. In order to enable
cached statement support you must switch enable the batched
statement option (FLAG_MULTI_STATEMENTS,
67108864, or Allow multiple
Of course. Please read through the default configs and the getting started
guide and xml dialplan information on the wiki.
Mike
This is of interest to me as well, would that be something like this:
extension name=ext_100_vm
condition field=caller_id_number expression=^100$/
On Thu, Nov 26, 2009 at 9:53 PM, Michael Jerris m...@jerris.com wrote:
http://dev.mysql.com/doc/refman/5.1/en/connector-odbc-news-3-51-18.html
MySQL Connector/ODBC now supports batched statements. In order to enable
cached statement support you must switch enable the batched
condition field=destination_number expression=^${caller_id_number}$
On Thu, Nov 26, 2009 at 4:36 PM, Joseph L. Casale jcas...@activenetwerx.com
wrote:
Of course. Please read through the default configs and the getting
started guide and xml dialplan information on the wiki.
Mike
This is
condition field=destination_number expression=^${caller_id_number}$
Of course:) Thank you!
jlc
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freeswitch list wrote:
condition field=destination_number expression=^${caller_id_number}$
I knew this day would come. After the accumulation of all of the knowledge from
the list members, the list has finally achieved sentience and is now answering
questions by itself. :-)
--
Russell
I tried now with phones directly attached to the freeswitch (without an
OpenSIPS in between). I also added the alias. But the behaviour is as
before:
No notify message from freeswitch, neither after register nor after a
voicemail is recorded.
Best regards
Peter
Brian West schrieb:
Yes an alias
LOL thats funny.
freeswitch, what is the meaning of life?
On Nov 26, 2009, at 4:45 PM, Russell Mosemann wrote:
freeswitch list wrote:
condition field=destination_number expression=^$
{caller_id_number}$
I knew this day would come. After the accumulation of all of the
knowledge from
On 11/26/2009 6:02 AM, Otis wrote:
Can I get FS to re-route incoming-calls to PSTN. If this has been
raised before could someone direct me to URL or link please
Since I don't have a hard line, I do something like:
include
extension name=2800
condition field=destination_number
On Wed, Nov 25, 2009 at 11:21:25AM -0700, Adam Ford wrote:
Awesome, thanks Andrew, I will have to keep an eye out for that patch.
Here's my patch in its (probably) final form.
http://eagle.bsd.st/~andrew/mweek2.diff
It includes a usage example that covers all but 2 of the US federal
holidays
Weird don't know how that got set to freeswitch list.
On Thu, Nov 26, 2009 at 9:27 PM, Rob Forman rob4manh...@gmail.com wrote:
LOL thats funny.
freeswitch, what is the meaning of life?
On Nov 26, 2009, at 4:45 PM, Russell Mosemann wrote:
freeswitch list wrote:
condition
Thanks for all the help, here is what I put in the dialplan, I tested
this and it is working for me.
!-- User calling Self Goes To voicemail --
extension name=usertoselfvmain
condition field=destination_number
expression=^${caller_id_number}$
action
Hi MC,
I have created won sample application yesterday, It was working fine. Today,
I checked that my local ip has changed. so, I changed the domain(IP) name in
sip-account settings in my x-lite configuration. After that x-lite is not
able to register with FS. I am getting error like
Ok. I've been running this system since FS was a beta. It stopped working
after a update.
I'll switch to a single profile. What NAT settings should it have? I really
want to get rid of the RECOVERY_ON_TIMER_EXPIRE error.
On Thu, Nov 26, 2009 at 9:44 PM, Michael Jerris m...@jerris.com wrote:
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