Hi,
Is it possible to originate calls from specific A500 ports with FreeSWITCH?
I am using a A504 (8 BRI interfaces), and I want some outbound calls to be
made from specific BRI interfaces.
I tried to modify OpenZAP config as follows:
conf/openzap.conf
[span wanpipe boostbri1]
trunk_type = bri
Sorry I hit 'send' by mistake...
Hi,
Is it possible to originate calls from specific A500 ports with FreeSWITCH?
I am using a A504 (8 BRI interfaces), and I want some outbound calls to be
made from specific BRI interfaces.
I tried to modify OpenZAP config as follows:
conf/openzap.conf
[span
Hello,
On Tue, Sep 1, 2009 at 1:06 PM, Mathieu Parentmath.par...@gmail.com wrote:
Hi,
On Fri, Aug 28, 2009 at 7:01 PM, Steve Underwoodste...@coppice.org wrote:
(snip)
The log shows the same thing happening every time. A bad CRC from the
far end, followed by a good DCS frame followed by
Hello *,
the latest bugfixes for luarun (pool allocation) fixed it.
It's working now with luarun (friday-monday) and bgapi (monday-today).
The last test with the new operator for sched_api is currently running.
Thx!
Beni.
attachment:
Hi,
Owe to the network bandwidth limitations (running on cellular phones
ip link) we are using speex/8000 as our voice codec.
However, when both parties are using that codec the sound is not to be
heard on the caller side.
looking at the log dumps one can see that
a) at the caller side, it
Hi,
Owe to the network bandwidth limitations (running on cellular phones
ip link) we are using speex/8000 as our voice codec.
However, when both parties are using that codec the sound is not to be
heard on the caller side.
looking at the log dumps one can see that
a) at the caller side, it
This looks and sounds like a case where pjsip isn't listening to our
SDP. If we 200 OK with speex on 102 and the far end starts sending it
on 98 then I suspect the client is broken if I'm not mistaken.
/b
On Sep 9, 2009, at 6:19 AM, Tzury Bar Yochay wrote:
Hi,
Owe to the network
Hi there,
the internal.xml and external.xml examples are for situations where FS is
running inside a company's private network, behind a NAT router. So
internal.xml connects the clients to FS without crossing a NAT, within the
same private network, while external.xml connects SIP providers
Hi all!
Is there any difference between auto_hunt=True and execute_extenstion?
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So if you have an extension name that is testing and the
destination number is testing then if testing is at the bottom of
the dialplan auto_hunt will make it warp right to it.
/b
On Sep 9, 2009, at 8:29 AM, Max Ivanov wrote:
Hi all!
Is there any difference between auto_hunt=True and
Those configs will still work.
/b
On Sep 9, 2009, at 6:16 AM, Jörg Hartmann wrote:
Hi there,
the internal.xml and external.xml examples are for situations where
FS is running inside a company's private network, behind a NAT
router. So internal.xml connects the clients to FS without
So if you have an extension name that is testing and the
destination number is testing then if testing is at the bottom of
the dialplan auto_hunt will make it warp right to it.
Ah, I see. Would it be correct to say that auto_hunt is similar to
goto and execute_extenstion behave like include ?
the instructions said build latest trunk.
did you actually do that? because lines of code in this gcore file do not
correspond to current trunk
which is why I asked you to update to it first.
Did you just rebuild 1.0.4 again? If you did rebuild trunk what version was
it?
we can't fix problems on
I have phones registered internally and can call among them. However, when
I dial 711 from an internal phone, freeswitch replies with 484 Address
Incomplete with reason INVALID_NUMBER_FORMAT. At the server console, I
see the following error:
[ERR] mod_sofia.c:2645 Invalid Gateway
Does
Because you named your gateway 192.168.72.253, not mediant1000.
You could name it mediant1000 and set param name=proxy
value=192.168.72.253 /, or use sofia/gateway/192.168.72.253/...
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
Dear All,
I'm looking for document,example for /filter command.
where to get it ?
BG
Dome C.
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This looks and sounds like a case where pjsip isn't listening to our
SDP. If we 200 OK with speex on 102 and the far end starts sending it
on 98 then I suspect the client is broken if I'm not mistaken.
/b
Could be, anyhow, note that this happens only both side using
speex/8000. If one party
I think the problem here is that the loader only keeps this method in scope
until completion then it drops the remoted connection. Therefore you should
not use threads in this method. Michael please correct me if I am wrong
here.
As an example of the failure simply just put a Sleep(1) call
I have a problem. After 10-20 minutes of Skype talk via cordless phone
connected to ATA the latter erroneously generated DTMF 'D' event.
Then skypiax looses connection while the call remain active in Skype
client. The only way to terminate it is to ask another party to hang
up:
(...)
2009-09-09
On Wed, Sep 9, 2009 at 10:17 AM, Dome Charoenyost d...@tel.co.th wrote:
Dear All,
I'm looking for document,example for /filter command.
where to get it ?
This is a handy way to add filters to what you see on the fs_cli. Event
sockets allow for filters and the /filter command lets
kernel32.dll!77e4bef7()
Here's that call stack.
[Frames below may be incorrect and/or missing, no symbols loaded for
kernel32.dll]
kernel32.dll!77e4bef7()
msvcr80.dll!78158e89()
mscorwks.dll!79e7a17a()
mscorwks.dll!79ea0fa8()
mscorwks.dll!79ea0eff()
mscorwks.dll!79e976cc()
On Wed, Sep 9, 2009 at 1:36 PM, Michael Collins m...@freeswitch.org wrote:
On Wed, Sep 9, 2009 at 10:17 AM, Dome Charoenyost d...@tel.co.th wrote:
Dear All,
I'm looking for document,example for /filter command.
where to get it ?
This is a handy way to add filters to what you
On Tue, Sep 8, 2009 at 1:20 PM, Nikolai Geordzhev n.geordz...@gmail.comwrote:
I`ve already tried the legs variable in cdr_csv.conf.xml, I have also tried
to use the loopback endpoint and to bridge the call to the internal
interface(so it can go out and in again generating the 2cdr-s I need)
A new discovery:public bool Load()
{
ThreadPool.QueueUserWorkItem((o) =
{
Log.WriteLine(LogLevel.Notice, Thread Starting. );
EventConsumer con = new EventConsumer(all, );
while (true)
{
On Tue, Sep 8, 2009 at 8:59 PM, Rogelio Perez rogelio.pe...@gmail.comwrote:
Hi guys,
My FS setup was working smoothly with mod_opal enabled until I had to
rebuild everything from scratch.
Now I have compiled everything following the same procedure (I even
have a script for that) and
What is the output of oz list and oz dump? Put them in
pastebin.freeswitch.org and link here in the mailing list.
-MC
On Tue, Sep 8, 2009 at 11:31 PM, Vassil Panayotov panayotov...@gmail.comwrote:
Sorry I hit 'send' by mistake...
Hi,
Is it possible to originate calls from specific A500
Yeah I noticed that but the thread was still terminating after a few seconds
anyway for me. Does it stay running for you?
Josh Rivers-2 wrote:
A new discovery:public bool Load()
{
ThreadPool.QueueUserWorkItem((o) =
{
Hi,
Is it possible to originate calls from specific A500 ports with
FreeSWITCH?
I am using a A504 (8 BRI interfaces), and I want some outbound calls to be
made from specific BRI interfaces.
Hello Vassil,
Unless you are using openzap trunk (and that probably means FreeSWITCH trunk
as
It does from a fresh start of FreeSWITCH. I've noticed, although not really
confirmed, a race condition between the unload and reload of managed code.
It seems that threads started in the newly submodule are terminated along
with the threads for the old, unloading submodule. Is that what you are
On Wed, Sep 9, 2009 at 01:20, Vassil Panayotov panayotov...@gmail.com wrote:
Hi,
Is it possible to originate calls from specific A500 ports with FreeSWITCH?
I am using a A504 (8 BRI interfaces), and I want some outbound calls to be
made from specific BRI interfaces.
You can't define several
The ILoadNotifcationPlugin is run in the appdomain created for the plugin, so
it should only get unloaded when the plugin gets reloaded. Spawning threads
here should work, it's definitely the intention that if you need a long-running
process, you can fire it up on load and have it work.
As to
I have a new thought on the crashes...I'm able to crash FreeSWITCH any time
I like, just by having an exception in a thread.
public class CrashFreeSWITCH : ILoadNotificationPlugin
{
public bool Load()
{
ThreadPool.QueueUserWorkItem((o) = { throw new
That's by design. If a thread fails, and there's no handler, then the
application could be in a corrupted state, so the CLR takes down the process.
I think there is a .NET 1.0 compat switch you can enable in the config if you
like exceptions to be silently ignored :).
-Michael
From:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello all!
I am having the following issue.
When I dial out over a FXO port (analog) for the first 8-10 seconds I
get no audio. If I wait I will eventually hear something. On inbound
calls audio works great in both directions. I used
The question is whether the CLR should take down the whole phone server due
to an unhandled exception...definitely the CLR should terminate...but
shouldn't it just log the exception to the console, not crash the core?
On Wed, Sep 9, 2009 at 6:30 PM, Michael Giagnocavo m...@giagnocavo.netwrote:
It should be the same, except using php syntax instead of perl.
Mike
On Sep 2, 2009, at 11:43 AM, Dome Charoenyost wrote:
2009/9/2 Michael Collins m...@freeswitch.org:
Are you trying to get a channel variable or capture DTMF input from
the
caller?
i try to make IVR by php outbound
HI,
I'm newbie in FS, I want to know how to implement h extension of asterisk to
FS. As I listed down below;
h =
{
NOOP(Call Completed with Carrier ${CARRIER});
goto add_cdr|h|1;
};
My other question is, which application/function/class is use in mod_perl to
check the
You can use a phrase macro but I am not sure that we set the position
in a way that you can expand it for the macro.
Mike
On Sep 1, 2009, at 10:37 AM, Dome Charoenyost wrote:
Dear sir,
I want to say posision in queue to caller but
fifo_chime_list can't say more than one sound
somewhere in that mess of my commands solaris is not liking
something. I tested this a lot on solaris and had it working on every
box i was in, so not sure what this could be. If you can get me into
a box in this state via ssh I can take a look.
Mike
On Sep 3, 2009, at 6:42 AM, Bruce
You should be able to handle hangups in one of two ways:1) Register a hangup
handler in your script or dialplan. This will execute a script on the hangup
of the call.
2) Use the Event Socket Layer(ESL) to listen to hangup events and then
perform your actions there.
You can find more about these
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