The dialplan is very simple:
extension name=Recording test
condition field=destination_number expression=^11\d*$
action application=sleep data=3000 /
action application=answer/
action
TC
TCit is gonna be easier to track.
TC
TCTomorrow i will test on 1.0.4 but please lets move to trunk.
i make it a bit later, to move tickets to jira and source to svn i
need some time to undertand how this system is works, especially jira.
audio issue is better now :)
however i have a
btw you are back with an old issue:
static const char modulename[] = h323;
static const char* h323_formats[] = {
G.711-ALaw-64k, PCMU,
G.711-uLaw-64k, PCMA,
GSM-06.10, GSM,
MS-GSM, msgsm,
SpeexNarrow, speex,
___
FreeSWITCH-users
Thanks to c6burns on IRC channel for the tip to use execute_on_answer in
combination with eval, and of course everyone here that pointing me to the
right direction.
I was able to execute sched_api with eval, but not with the combination of
execute_on_answer. The argument just don't get parsed as
Yeah the word **hearbeat** was confusing me, now its ok.
So one use case would be to count live calls with much detailed information.
is it *resource*-*intensive** operation? can i do it on wholesale traffic
(500+ calls).*
...any other interesting use cases?
Thanks.
On Thu, Oct 22, 2009 at
On 2009-10-23 10:16 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
TC TC
TC TCit is gonna be easier to track.
TC TC
TC TCTomorrow i will test on 1.0.4 but please lets move to trunk.
TC
TC i make it a bit later, to move tickets to jira and source to svn i
TC need some time to
On 2009-10-23 10:44 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
TCbtw you are back with an old issue:
TC
TC
TCstatic const char modulename[] = h323;
TCstatic const char* h323_formats[] = {
TCG.711-ALaw-64k, PCMU,
TCG.711-uLaw-64k, PCMA,
TCGSM-06.10, GSM,
TCMS-GSM,
i meant you switched PCMA and PCMU...
T.
2009/10/23 Georgiewskiy Yuriy bottle...@icf.org.ru
On 2009-10-23 10:16 +0200, Tihomir Culjaga wrote
freeswitch-us...@lists.fre...:
TC TC
TC TCit is gonna be easier to track.
TC TC
TC TCTomorrow i will test on 1.0.4 but please lets move to trunk.
On 2009-10-23 13:22 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
my bad, thx.
TCi meant you switched PCMA and PCMU...
TC
TCT.
TC
TC2009/10/23 Georgiewskiy Yuriy bottle...@icf.org.ru
TC
TC On 2009-10-23 10:16 +0200, Tihomir Culjaga wrote
TC freeswitch-us...@lists.fre...:
TC
TC TC
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Anthony,
thanks! The unknown thing is gone as long as you don't use
originate_callee_id_name
The originate_callee_id_name chvar causes caller's AND callee's
display (caller name line)to be updated with the content of that chvar
as soon as you
TC3. can we control mediaWaitForConnect flag within setup message via
config
TCfile setting?
what is mediaWaitForConnect flag, may be another trmin in h323?
If the calling endpoint sets the mediaWaitForConnect element to TRUE in the
Setup message, then
the called endpoint shall not send
-BEGIN PGP SIGNED MESSAGE-
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Hello,
when I use pre_answer app and right after that the send_display app,
there is no INFO message anymore.
My dialplan condition:
condition field=${sip_hangup_phrase} expression=^Do Not Disturb
break=on-true
action
On 2009-10-23 13:52 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
TC
TC TC3. can we control mediaWaitForConnect flag within setup message via
TC config
TC TCfile setting?
TC
TC what is mediaWaitForConnect flag, may be another trmin in h323?
TC
TC
TC
TCIf the calling endpoint sets
TCcheck H225_Setup_UUIE H323SignalPDU::BuildSetup within
src/h323pdu.cxx
TC(H323plus)
i think it can be implemented later, but, why it may be needed? can you
explain some
situation where it need?
TC
TCyou should handle this and postpone pre_answer until you get an open LC.
On 2009-10-23 14:38 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
TC
TC TCcheck H225_Setup_UUIE H323SignalPDU::BuildSetup within
TC src/h323pdu.cxx
TC TC(H323plus)
TC
TC i think it can be implemented later, but, why it may be needed? can you
TC explain some
TC situation where it
Hello,
in a freeswitch cluster (FS1 and FS2) behind an OpenSIPS I want
Freeswitch to register to external gateways through the OpenSIPS load
balancer, in order to later receive incoming calls through the load
balancer.
Is there a way to tell Freeswitch in it's Gateway definition to define
an
TC
TCbool FSH323Connection::OnReceivedProgress(const H323SignalPDU pdu)
TC{
TCPTRACE(4,
mod_h323\t==FSH323Connection::OnReceivedProgress);
TC
TCPTRACE(4, mod_h323\t==FSH323Connection::OnReceivedProgress
-
TCdisabled pre_answer);
TC
TC
Hi
I am new to Freeswitch so my question may be a stupid question.
I just want to know how to disable the personal greeting to the default one.
One user has recorded his personal greeting now how can he make this default.
I could not find any option for the same.
Plz advice.
Thanks regards
hello, all,
I want bridge call to gateway without register.
Example:
action application=bridge data=sofia/sipinterface_1/$
1...@192.168.111.101:5060/
Error is : sofia_reg.c:1568 No Matching gateway found
My purpose is the gateway is realtime changed from script(etc lua,
There was a concern with letting this happen before answer.
I changed it in last revision, try it out.
The problem is that the way polycom does it, it violates the RFC.
Also know you can set sip_callee_id_name and sip_callee_id_number before you
send a ring_ready pre_answer or answer to send the
Hello:
Your issue is that this portion of the above string sipinterface_1 has to
be a valid sip profile. To do this:
-- determine the appropriate sip profile you want to use (usually
external)
-- send the traffic to whatever ip you want via this portion of your
string
a solution to H323 endpoint = FS = SIP user no audio issue
is to disable a wait for tx Audio ... for case
SWITCH_MESSAGE_INDICATE_ANSWER:{
//m_txAudioOpened.Wait();
case SWITCH_MESSAGE_INDICATE_ANSWER:{
switch_log_printf(SWITCH_CHANNEL_LOG,
it's probably related to escaping the data.
I was sick of watching you suffer so i added api_on_answer variable to
trunk.
On Fri, Oct 23, 2009 at 3:54 AM, Henry Huang red.rain.se...@gmail.comwrote:
Thanks to c6burns on IRC channel for the tip to use execute_on_answer in
combination with eval,
what you get if you run
sofia status profile sipinterface_1
- SAT
On Fri, Oct 23, 2009 at 4:58 PM, Shelby Ramsey sicfsl...@gmail.com wrote:
sipinterface_1
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
On 2009-10-23 16:57 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
i have question to developers about one proce in fs
src/switch_ivr_originate.c
static switch_status_t
originate_on_consume_media_transmit(switch_core_session_t *session)
{
switch_channel_t *channel =
if you were on trunk that line of code would be gone.
you really can't do development on 1.0.4 its 6 months old and it will cause
you more trouble than you think when you eventually upgrade if you do not do
it soon.
2009/10/23 Georgiewskiy Yuriy bottle...@icf.org.ru
On 2009-10-23 16:57 +0200,
should be even better in 15210
On Fri, Oct 23, 2009 at 6:35 AM, Helmut Kuper helmut.ku...@ewetel.dewrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Anthony,
thanks! The unknown thing is gone as long as you don't use
originate_callee_id_name
The originate_callee_id_name chvar
On 2009-10-23 10:37 -0500, Anthony Minessale wrote freeswitch-us...@lists.f...:
i have no way to install trunk at this time, i will go out of hospital about
one week later, after
this i will can try it on trunk.
AMif you were on trunk that line of code would be gone.
AMyou really can't do
FYI,
The weekly call will begin soon. The agenda is here:
http://bit.ly/O0oGB
Note that the wiki is giving me problems with editing. I will let you know
when Raymond and I get it squared away.
-MC
___
FreeSWITCH-users mailing list
2009/10/23 Georgiewskiy Yuriy bottle...@icf.org.ru
On 2009-10-23 10:37 -0500, Anthony Minessale wrote
freeswitch-us...@lists.f...:
i have no way to install trunk at this time, i will go out of hospital
about one week later, after
this i will can try it on trunk.
AMif you were on trunk
Any ideas on this one. Look slike only way rite now is to have a different Dest
Phone number for a moderator.
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Ujjval
Karihaloo
Sent: Thursday, October 22, 2009 9:02 PM
To:
I think we can't create two profiles per single IP. So what is the
workaround to solve this issue?
SAT
On Tue, Oct 20, 2009 at 7:28 AM, br...@freeswitch.org
br...@freeswitch.orgwrote:
It's because Sofia used one thread per profile so depending on load you
will see one CPU used more.
/b
The first caller isn't challenged for the pin (don't really know why--
maybe somebody else can elaborate on how the pin is designed to be
used). So to work around it, I validate the conference pin and
moderator pin independently via an IVR (or dynamically with a script
and odbc call) then
I installed FS on a machine with a Sangoma A101D (PRI) card and if I make an
inbound call to the FS IVR, it does not recognize DTMF digits from the PSTN
phone. If I call IVR from an internal phone, then it does recognize the
DTMF digits. I have mostly default configurations for everything.
Well first off you're not defining a pine here...
confn...@profilename+flags{mute|deaf|waste|moderator}+[conference pin
number]
That might be why its not asking for a pin.
/b
On Oct 23, 2009, at 12:30 PM, Rob Forman wrote:
entry action=menu-exec-app digits=1 param=conference
poor bbhenry :)
Added in r15207 please test and update docu if necessary:
http://wiki.freeswitch.org/wiki/Variable_api_on_answer
On October 23, 2009 11:02:07 am Anthony Minessale wrote:
it's probably related to escaping the data.
I was sick of watching you suffer so i added api_on_answer
Right- this was an example without using the built-in pin but instead
using IVR.
On Oct 23, 2009, at 12:43 PM, Brian West wrote:
Well first off you're not defining a pine here...
confn...@profilename+flags{mute|deaf|waste|moderator}+[conference pin
number]
That might be why its not
Hello everyone,
I'm having some issues with SIP and TCP. I've used it before with
success but I'm seeing some strange behavior...
Level 7 debugs with siptrace on both profiles. UDP invite from
softphone comes in on port 5062, it's supposed to bridge to
10.70.0.62. When configured to use
On 2009-10-23 18:26 +0200, Tihomir Culjaga wrote freeswitch-us...@lists.fre...:
TC2009/10/23 Georgiewskiy Yuriy bottle...@icf.org.ru
TC
TC On 2009-10-23 10:37 -0500, Anthony Minessale wrote
TC freeswitch-us...@lists.f...:
TC
TC i have no way to install trunk at this time, i will go out of
I just re-tested with the pin in my dial plan:
action application=conference data=conference 123...@default
+flags{}+1234 /
And it doesn't challenge me for the pin. I just drop right in. I
figured this is how it was intended, since the wiki says the pin is
set initially and only
On Fri, Oct 23, 2009 at 10:16 AM, Saeed Ahmad saeedahmad1...@gmail.comwrote:
I think we can't create two profiles per single IP. So what is the
workaround to solve this issue?
SAT
Two profiles can share the same IP address, just not the same IP and port.
The example internal and external
On Fri, Oct 23, 2009 at 12:36 AM, Maciej Aniserowicz
maciej.aniserow...@gmail.com wrote:
The dialplan is very simple:
extension name=Recording test
condition field=destination_number
expression=^11\d*$
action application=sleep data=3000 /
I have mod xml_curl installed and I am getting the following passed to my
script.
[hostname] = myhost.local
[section] = dialplan
I also have multiple versions of FS running on the same box. Is there a way
to have each FS instance on my box have a unique hostname ?
Thanks in advance.
Can't you use different contexts or something else to tell them apart?
On Fri, Oct 23, 2009 at 3:34 PM, freeswitch noob
freeswitch.n...@gmail.com wrote:
I have mod xml_curl installed and I am getting the following passed to my
script.
[hostname] = myhost.local
[section] = dialplan
I also
one real quick way would be put different GET var in each server's binding
On October 23, 2009 03:46:11 pm Kristian Kielhofner wrote:
Can't you use different contexts or something else to tell them apart?
On Fri, Oct 23, 2009 at 3:34 PM, freeswitch noob
freeswitch.n...@gmail.com wrote:
I
Yeah, I was just trying to make it easier on myself. I have scripts from a
friend that parse xml_curl requests based on the hostname, I was hoping to
not have to re-write them to read something else from the post that FS makes
from xml_curl. But from what it sounds like I will have to.
On Fri,
Why not simply overwrite the value of the variable used throughout the
script...
-- xml_curl.conf --
...
param name=gateway-url
value=http://localhost/index.php?xhostname=myhost; bindings=dialplan/
...
-- index.php --
?
$_REQUEST['hostname'] = $xhostname;
...
- Original Message
Please note that this would essentially be taking Chris' suggestion a little
further but the effort involved would be minimal.
- Original Message -
From: Metik
To: freeswitch-users@lists.freeswitch.org
Sent: Friday, October 23, 2009 5:37 PM
Subject: Re: [Freeswitch-users]
Forget it--this will not work because FS uses a POST (vs GET). Most likely it
would attempt to actually POST to a file named index.php?xhostname=myhost.
However, if there was some way to add an arbitrary POST variable to the HTTP
transaction, it would work. I am sure the FS developers have
Fake the hostname in the query string:
http://test/dir.php?hostname=mmmpotatoes
Add this to the top of your buddy's script (if its even PHP):
$hostname = $_POST['hostname'] = $_REQUEST['hostname'] = $_GET['hostname'];
On October 23, 2009 06:18:41 pm Metik wrote:
Forget it--this will not work
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