Hi Klaus,
Some Perl code snippets - we use:
call_command(bridge, sofia/gateway/bt/$ntd);
which, in turn, is:
sub call_command($$) {
my $cmd = shift;
my $arg = shift;
print $sock sendmsg\ncall-command: execute\nexecute-app-name:
$cmd\nexecute-app-arg: $arg\n\n;
}
Cheers
Hi,
I am making a simple bridge between two call legs :
Client --(a-leg)-- FS --(b-leg)--Provider
How can I get information like network-address of the Provider,
media-address,
port used, media port used etc. from the second leg (b-leg)?
Is all the information provided by the a-leg
2 options.
1) enable b-leg logging on the cdr module.
2) you can use the prefix bleg_ in a variable context to get to
caller_profile members
from the b leg.
eg ${bleg_caller_id_name}
On Wed, Dec 3, 2008 at 7:30 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
I am making a simple
b-leg logging is enabled in the cdr module. but in the cdrs I cannot get
any variables that refer to the b-leg.
I tried the second way using ${sip_to_host} and {bleg_sip_to_host} but :
a) the variable returns the FS IP on the a-leg CDR (correctly)
b) the variable returns nothing on the b-leg
Hi All,
Thanks for your feedback. I must be doing something fundamentally wrong.
Inbound socket is working without problems. But the exact things that i do on
inbound socket, i'm not able to replcate them on outbound socket.
The global picture: I have on Xlite registered at extension 1002 and
Hi,
*I have newly installed freeswitch in another machine.
**After starting the freeswitch I try to dial a extension from console but
when i call extension 1002 from freeswitch console, call is connected to
extension 1002, but FS is aborted but call is established in1002.*
*When i dial from
I looked in the b-leg xml cdr and the ip address is not there (for
signaling) it is only there
for media (${remote_media_ip}) which is not the same thing now, is it?
While we are at it, I noticed that the ${local_media_port} and
${remote_media_port}
have the same value for each CDR (a or b
2. You don't need to send the UUID in after the sendmsg - FS already
knows which call you're controlling.
Bingo! That was it.
Thanks,
Klaus.
--
Sensationsangebot verlängert: GMX FreeDSL - Telefonanschluss + DSL
für nur 16,37 Euro/mtl.!* http://dsl.gmx.de/?ac=OM.AD.PD003K1308T4569a
It's not an unreasonabe request so i added a patch you can test for me to
trunk that sets network_addr on the reciept of a reply to an invite on an
outbound call. and the 2 variables sip_reply_host and sip_reply_port
local and remote media port reflects the port being used between that leg
and
please clean all the core.* files
reproduce the problem which will generate a core.xyz file (xyz is some
number)
run the command.
gdb /usr/local/freeswitch/bin/freeswitch core.xzy
when it loads issue the command
bt
and send me the output.
--
Anthony Minessale II
FreeSWITCH
Hi everyone,
I am Gab and just joined the group. Also, I am new to FS but want to learn
and delve into the dept as fast as possible. I have one last mile question
and was hoping I could pick from someone's wealth of knowledge and
understanding of the platform.
I have setup FS with 5 extensions
I'll try the patch. Thank you for your time.
As for the local and remote media ports :
I have an endpoint with IP xxx.xxx.xxx.xxx and an FS box with IP
yyy.yyy.yyy.yyy.
In a SIP bridge each side of the call leg between the two boxes will
pick a udp port in order to send/receive traffic.
In my
Hello everybody,
I am new to the list and I hope I can find some help here, regarding an
issue I am experiencing with the CDRs written by Freeswitch.
The thing is, I am using the max-sessions and the sessions-per-second
parameters in switch.conf.xml to limit the maximum number of simultaneous
On Wed, Dec 3, 2008 at 10:27 AM, Lachezar Valchev
[EMAIL PROTECTED] wrote:
Hello everybody,
I am new to the list and I hope I can find some help here, regarding an
issue I am experiencing with the CDRs written by Freeswitch.
The thing is, I am using the max-sessions and the
From: Michael Collins [EMAIL PROTECTED]
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Wed, 3 Dec 2008 11:41:04 -0800
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] CDR generated on maximum sessions reach
On Wed, Dec 3, 2008 at 10:27 AM, Lachezar Valchev
[EMAIL
looks like a typo in the code. I guess nobody ever looked at that field
before.
it should be fixed in r10582
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:[EMAIL PROTECTED] [EMAIL PROTECTED]
GTALK/JABBER/PAYPAL:[EMAIL
Hi Gab!
Welcome to FreeSWITCH. Thanks for your questions. I'm trying to learn all of
this stuff and help others so bear with me while I research these and help
you find the answers.
BTW, are you on IRC? you can visit us for realtime help, #freeswitch on
irc.freenode.net
-MC (mercutioviz on irc)
Is these apps will work for you?
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect
On Sat, Nov 29, 2008 at 6:18 AM, Barray McKee [EMAIL PROTECTED] wrote:
Hello,
I am implementing 2 load balancing FS behind a pair of
And thank you for testing and being gracious! :)
-MC
On Wed, Dec 3, 2008 at 1:24 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I tested both patches from the trunk : network_addr is set to the remote
IP on the b-leg
and local media port and remote media port hold the correct values when
On Tue, Dec 2, 2008 at 8:18 PM, Michael Collins [EMAIL PROTECTED] wrote:
Kristian,
Are you on the IRC channel by any chance?
-MC (IRC: mercutioviz)
Me? Never!
--
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star.com
Hi
I was trying to create extension in which I would check whether
privacy=full in Remote-Party-ID header is set.
So I made this.
condition field=${screen_bit} expression=^true$ break=never
action application=set
data=origination_caller_id_number=Anonymous/
/condition
But
The screen bit is a trust bit... ie: do we trust the RPID we got from the
upstream or not
K
From: Piotr Sobolewski [EMAIL PROTECTED]
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Wed, 3 Dec 2008 23:33:24 +0100
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users]
You scared? muhahaha Will I see you at ClueCon 09, its the first week
in Aug. again in Chicago.
/b
On Dec 3, 2008, at 4:26 PM, Kristian Kielhofner wrote:
Me? Never!
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Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
I think we covered this on IRC already didn't we?
/b
On Dec 3, 2008, at 4:33 PM, Piotr Sobolewski wrote:
Is it a bug or me doing something wrong?
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
On Thu, Dec 4, 2008 at 1:14 AM, Ken Rice [EMAIL PROTECTED] wrote:
The screen bit is a trust bit... ie: do we trust the RPID we got from the
upstream or not
K
I had privacy_hide_number in cdr_csv and I was thinking it was
screen_bit, all that confused me.
Now I understand where I was wrong.
I already told you this one on IRC too :P email is too slow today :)
/b
On Dec 3, 2008, at 8:02 PM, Piotr Sobolewski wrote:
BTW: is there a way to remove RPID header?
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Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
Hi guys,
I've a strange issue with FS , version svn -r10584 ,
when FS bridges a call first 3 seconds of audio are missing , looks that
only happens on PSTN calls and using ringback or transfer_ringback. This
only happens in calls from PSTN , not from VOIP. Some scenarios i tried
to isolate
what does PSTN represent?
I know what the PSTN is but how are you reaching it?
is it TDM, SIP etc... what gateway type other details.
On Wed, Dec 3, 2008 at 7:03 PM, Angel Carpintero [EMAIL PROTECTED] wrote:
Hi guys,
I've a strange issue with FS , version svn -r10584 ,
when FS bridges a
No TDM , all is SIP :
PSTN --- Sip Proxy_A -- FS ( brigde ) ringback/transfer_ringback
- Sip Proxy_B -- PSTN
In logfile i think you can get some details about Media Gateways
( Sonus ) PSTN inbound / outbound is provided by Level3.
I can get a capture of a call if you want, in capture the
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