Thanks for reply, well actually I'm doing billing after call hangup. If h
extension is interupts I'm sending to it to addcdr context which interupts
perl script for billing purpose. As I'm listing down below asterisk
configuration;
h =>
{
NOOP("Call Completed with Carrier ${CARRIER}");
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
MC,
Rhino ended up not needing to reflash it. I am trying to sort out why
the driver is segfaulting with Zaptel 1.4.12.1. One of the guys at
Rhino has a FS box with the same card in it. I am going to try to
replicate his settings tomorrow.
You'll get this for transfers, and each leg... Can you elaborate on
the call scenario more?
/b
On Sep 10, 2009, at 3:47 PM, email lists wrote:
Hello,
Using the mod_radius_cdr module w/ Freeswitch, I am seeing duplicate
RADIUS messages being generated for individual calls (sample
message
There will NEVER be an easy way to do billing inline and we never
support such crazy because its WRONG. That said... You're not the
first to approach FreeSWITCH with an Asterisk mentality.
Billing inline = NO
Billing direct to DB = NO
Post Processing = YES, do this.
Nibble/Heartbeat billing v
What puzzles me is why you can't just update to SVN trunk to get
uuid_exists in the first place? Its going to be 1.0.5 the more people
we have testing the faster that happens.
/b
On Sep 10, 2009, at 4:24 PM, Phillip Jones wrote:
> Strangely - the uuid_getvar uuid
> workaround does not wo
FYI,
We are on schedule for the weekly conference call. Please be sure to join
us! More information is available here:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_11
Talk to you all tomorrow!
-Michael
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Well, we have absolutely no idea what the background thread is doing. It might
be critical, and the fix is trivial: put a try/catch on it. This is the model
all .NET applications have. Background threads doing bad things should always
take down the process.
However, that's a good point about Lo
Hi,
we have a Freeswitch server on a public IP and a few phones behind
NAT. The phones are configured to use STUN and can register and call
each other fine.
The problem is that after attended transfer (using the mechanism the
phones provide - REFER) is finished, the two parties can't hear each
othe
As a sanity test, can you bring up a bridged call and run this test directly
from the command line and compare the results to what's in your script?
-MC
On Thu, Sep 10, 2009 at 2:24 PM, Phillip Jones wrote:
> Strangely - the uuid_getvar uuid workaround
> does not work for me.
>
> This is the re
nevermind, I think i fixed it in latest trunk, please test...
On Thu, Sep 10, 2009 at 12:53 PM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:
> could you produce a minimal offending script we could test with on our box?
>
>
>
> On Thu, Sep 10, 2009 at 12:43 PM, Nik Middleton <
> nik.mi
Strangely - the uuid_getvar uuid workaround does
not work for me.
This is the result of:
apiResult = fsApi.ExecuteString(string.Format("uuid_getvar {0} uuid",
call.Uuid));
Log.WriteLine(LogLevel.Alert, "RESULT: uuid_getvar {0} is: {1}", call.Uuid,
apiResult);
returns a different uuid?
2009
Hello,
Using the mod_radius_cdr module w/ Freeswitch, I am seeing duplicate
RADIUS messages being generated for individual calls (sample messages
for one call below). Looking at the "Acct-Unique-Session-Id" and
"Acct-Session-Id" fields, it would appear that perhaps each call leg
results in a p
On Thu, Sep 10, 2009 at 12:32 PM, Diego Viola wrote:
> Lets make sure we add it on the wiki too =D.
>
> Yep, as soon as we verify its functionality we'll wikify it. :)
-MC
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http://li
Is this the same card that had issues and Rhino had to re-flash?
-MC
On Wed, Sep 9, 2009 at 8:54 PM, Dan wrote:
> -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
Lets make sure we add it on the wiki too =D.
Diego
On Thu, Sep 10, 2009 at 7:03 PM, Dome Charoenyost wrote:
> 2009/9/11 Anthony Minessale :
> > I added a var for it (I hope properly) to r14806
>
> Thanks. I'll try
>
> Dome C.
>
> >
> >
> > On Thu, Sep 10, 2009 at 2:37 AM, Michael Collins
> wro
Would you mind creating a wiki page on this? It would be good to document
why you need this functionality, that is, what your business case is and how
this solution helps you resolve your issue. Create your own page and then
link to it from this page:
http://wiki.freeswitch.org/wiki/Examples
Thank
2009/9/11 Anthony Minessale :
> I added a var for it (I hope properly) to r14806
Thanks. I'll try
Dome C.
>
>
> On Thu, Sep 10, 2009 at 2:37 AM, Michael Collins wrote:
>>
>>
>> On Wed, Sep 9, 2009 at 11:32 PM, Dome Charoenyost wrote:
>>>
>>> 2009/9/10 Michael Jerris :
>>> > You can use a phras
On Thu, Sep 10, 2009 at 11:23 AM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:
> I added a var for it (I hope properly) to r14806
>
>
> Dome,
Look for the fifo_position chan var in the latest SVN. Please try it out
and see if it works, and make sure that it gets updated as the person'
I'm only concerned with the difference in treatment.
public class CrashFreeSWITCH : ILoadNotificationPlugin
{
public bool Load()
{
ThreadPool.QueueUserWorkItem((o) => { throw new
NotImplementedException(); });
return true;
}
}
Crashes the ent
I added a var for it (I hope properly) to r14806
On Thu, Sep 10, 2009 at 2:37 AM, Michael Collins wrote:
>
>
> On Wed, Sep 9, 2009 at 11:32 PM, Dome Charoenyost wrote:
>
>> 2009/9/10 Michael Jerris :
>> > You can use a phrase macro but I am not sure that we set the position
>> > in a way that
could you produce a minimal offending script we could test with on our box?
On Thu, Sep 10, 2009 at 12:43 PM, Nik Middleton <
nik.middle...@noblesolutions.co.uk> wrote:
> Yes I did a make current, and make sure,
>
>
>
> I’ve now reverted back to the latest release and all’s well.
>
>
>
> Is the
Yes I did a make current, and make sure,
I've now reverted back to the latest release and all's well.
Is there anything else I could try so as to ensure I've not got any
issue like you suggest?
Regards,
From: freeswitch-users-boun...@lists.f
Surely this is because you are calling ev.serialized_string and ev is null
if there is not event waiting?
i.e. con.pop(0); returns null if no event is waiting.
Therefore
if(ev != null)
Log.WriteLine(LogLevel.Notice, "Event: " + ev.serialized_string);
would work, alternatively calling
Here you have an example of how to get the CDR vars from the hangup event
and send it to the db.
http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/diegoviola/ruby/callcard/cdr.rb
Regards,
Diego
On Thu, Sep 10, 2009 at 8:06 AM, Ken Rice wrote:
> If you are just doing CDR processing the
Hi Nikolai,
Thanks for the response. Unfortunately I am already running without -
hp. Just plain ol' ./bin/freeswitch.
Also, I didn't mention it but I'm current and running off the build
(14805).
Rob
On Sep 10, 2009, at 11:58 AM, Nikolai Geordzhev wrote:
FYI, I got the same/similar iss
Well, that shouldn't happen, get a trace and open a jira or contact the
developers to resolve the issue.
Best regards,
Diego
On Thu, Sep 10, 2009 at 4:58 PM, Nikolai Geordzhev wrote:
> FYI, I got the same/similar issue with a 10-line perl script, starting FS
> without -hp option worked for me.
FYI, I got the same/similar issue with a 10-line perl script, starting FS
without -hp option worked for me.
On Thu, Sep 10, 2009 at 7:28 PM, Rob Forman wrote:
> Hi all,
>
> I wrote a small 10-line python script wrapping "txfax" (
> http://pastebin.freeswitch.org/10274
> ). Basically it originat
I wonder if maybe you have a build issue with an older mod_lua with a newer
FreeSWITCH
did you update via make current?
On Thu, Sep 10, 2009 at 11:11 AM, Nik Middleton <
nik.middle...@noblesolutions.co.uk> wrote:
> Hi Guys,
>
>
>
> I’m getting a core dump when running an lua script that’s been
Hi all,
I wrote a small 10-line python script wrapping "txfax"
(http://pastebin.freeswitch.org/10274
). Basically it originates a call with Session(), calls mod_fax txfax
application, then hangs up.
The weird thing is that this works fine the first I run it from
fs_cli. When I run a secon
Hi Guys,
I'm getting a core dump when running an lua script that's been fine for
months
In Freeswitch_lua.cpp line 92 is being called, but it's not clear what
exactly this is doing
lua_State *Session::getLUA()
{
if (!L) {
switch_log_printf(SWITC
Were both of them actually loading completely? As in, they had an app or api
plugin?
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Jeff Lenk
Sent: Thursday, September 10, 2009 8:34 AM
To: freeswi
Ummm bridge...
ie:
This will rewrite the request URI on an invite... However keep in mind the
FreeSWITCH is a B2BUA and not a proxy so this might not be exactly what you
are looking for
From: Carolina Benavides Cabrera
Reply-To:
Date: Thu, 10 Sep 2009 09:51:42 -0500
To:
Subject: [Freeswi
Hi.
I'm looking for any function similar to "rewritehostport" function of
OpenSIPS.
I need to route the SIP signalling toward Sailfin AS, replacing the
RequestURI.
Thanks for your help
Carolina
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Thanks Michael,
I was seeing some strange behavior in a static compiled dll versus
dynamically loaded and compiled scripts.
This code when used in a dll was blowing up when a thread was spawned in the
ILoadNotificationPlugin.Load but is not when used in a dynamically loaded
script(csx).
Code:
This does now...
/b
On Sep 10, 2009, at 4:52 AM, Jason White wrote:
>
> Yes, but by default the internal profile doesn't handle nat, which
> is why (if
> I recall correctly) it has been recommended that the external
> profile be used
> to register clients that are not on the local network wh
Hi,
I want to share my findings in making work Freeswitch as SBC when Kamailio is
doing serial forking. FS doesnt take any routing decision, it receives the
R-URI from Kamailio.
After the 1st route failed, I was receiving "482 Request merged" for the 2nd
route. That was because the SIP tran
> 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
Brian,
The original "a=rtpmap:98 SPEEX/8000" is generated by FS and NOT by the cli
Hi,
I need to have local_media_ip, local_media_port, remote_media_ip, and
remote_port to be available in the b-leg. Is there anyway to do that?
Since these variables are only created after the call has been answered, I
can't do http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBS
Hi!
Actually I think the problem was with the acl list..
I had put commented line below in. How did this cause the internal
profile be executed in the public extension?
<- when
removed it works again
Thanks
On Thu, Sep 10, 2009 at 10:09 PM, Tihomir Culjaga wrote:
> check your
check your sip profiles
/usr/local/freeswitch/conf/sip_profiles/external.xml
/usr/local/freeswitch/conf/sip_profiles/internal.xml
/usr/local/freeswitch/conf/vars.xml
It is simple :P
On Thu, Sep 10, 2009 at 1:39 PM, Mark Campbell-Smith <
mcampb
Hi!
Where in the dialplan does FS decide which context is used for
processing.. I am dialing an outbound call but the call is being
processed in context public and not default?
mod_dialplan_xml.c:315 Processing Extension1000-> in context public
Why is FS choosing the public context for an outbo
hi jason,
yes, we're aware of the external profile. but the sample profile shows only
how to register FS to SIP gateways - not external clients registering to FS.
the directory/default/*xml belongs to the internal profile. how can we
create another directory for external clients? we like to see sa
Nandy Dagondon wrote:
> for outside clients to register w/ the internal profile, the router has to
> forward port 5060 to FS. am i correct?
Yes, but by default the internal profile doesn't handle nat, which is why (if
I recall correctly) it has been recommended that the external profile be used
t
hi brian,
for outside clients to register w/ the internal profile, the router has to
forward port 5060 to FS. am i correct?
/nandy
On Wed, Sep 9, 2009 at 10:28 PM, Brian West wrote:
> Those configs will still work.
> /b
>
> On Sep 9, 2009, at 6:16 AM, Jörg Hartmann wrote:
>
> Hi there,
>
> the
On Wed, Sep 9, 2009 at 10:04 PM, Dmitry Bely wrote:
> 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 termina
If you are just doing CDR processing the easiest ways are using the event
socket to trigger this on the hangup event and you will get all the data you
want or use mod_xml_cdr which will either drop a file or fire a web request.
You can use either of these methods to trigger a billing update
From
FYI,
The Friday meeting agenda is posted here:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_11
Please add agenda topics as needed.
We invite all to join us this coming Friday at 11AM CST.
-Michael
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João Mesquita wrote:
> No can do. There are better tools to do that. tshark, wireshark and all
> other variants can do that for you.
I would recommend learning about read filters in tshark/wireshark, which
support a very flexible filtering language that is suitable for capturing SIP
traffic.
__
On Wed, Sep 9, 2009 at 10:16 PM, Ahmed Munir wrote:
> 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 ques
On Wed, Sep 9, 2009 at 11:32 PM, Dome Charoenyost wrote:
> 2009/9/10 Michael Jerris :
> > 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.
>
> fifo don't have queue position variable ?
>
>
I have a feeling that this functiona
2009/9/9 João Mesquita
> No can do. There are better tools to do that. tshark, wireshark and all
> other variants can do that for you.
>
>
JM is correct here. The filter is to filter FreeSWITCH events, not log
information hitting the console. I highly recommend this page for SIP
traces:
http://w
No can do. There are better tools to do that. tshark, wireshark and all
other variants can do that for you.
jmesquita
On Thu, Sep 10, 2009 at 3:29 AM, Dome Charoenyost wrote:
> How to use filter with sofia trace on ?
> Like Asterisk we can debug sip by
>
> sip set debug ip xx.xx.xx.xx.
>
> BG
Try to explain a little bit better what add_cdr does right here. Unlike
Asterisk, FreeSWITCH do have lots on information on CDR and it feels like
you are trying to do things on the wrong place.
If you want to understand where I am going with this, take a look at this
example XML CDR that can be po
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