Its works, cool!
Thanks Minessale again
and also thanks to those all related to development of FS
FS rocks!
Anthony Minessale-2 wrote:
the variables would be set on the b leg of the call not the a leg so you
would get them
from the cdr.
Since this is a pain for some people I added code
Hi again,
Now i found two pdds one for A leg and one for B leg
Both is different B leg's pdd is lower then A's pdd.
Will any body let me know how PDD is calculated for both legs in FS.
Thanks
shehzad p wrote:
Its works, cool!
Thanks Minessale again
and also thanks to those all related
However, I don't think we're at the point of how to rip out a legacy PBX
and drop in FS. Not yet, but give us some time... :)
Why not? You can simply use a Asterisk box for the Zap stuff. That is
how we do it. Works pretty well. Better still use E1/T1 gateways or POTS
gateways for analog
Thank you for your input. I am clearly naive about what may be required,
and I need to get more details on what is currently in use.
The environment is not 'can't-be-down-for-a-second' - it has crashed
before - but that became an emergency. It is possible to plan for short
( 1/2 hour )
Hi All,
Does anyone know how to stop freeswitch from messing with the callerid
(specifically name portion) of a transferred call out of a sip gateway?
In other words, all calls, once transferred out of the gateway appear to come
from the gateway name (or more specifically the username
Hi,
Thanks Anthony for your response.
Unfortunately removing the 'kick' didn't make a difference.
The result is also different if I change the order of the UUIDs around.
In one scenario I get [NORMAL_CLEARING], in the other
[DESTINATION_OUT_OF_ORDER].
I added the code (again) and 2 snippets of
Hi,
Just a thought ... is there a way in javascript to (re)create a session
object given an existing uuid?
If that was the case, I could do something like:
var session = new Session (uuid);
session.bridge(customer_service_url);
If that's not possible, I appreciate any thought about my current
OK, I got it. The replay vulnerability only happens when key exchange is
done via unencrypted SIP. I understand that with TLS the Invite message
cannot be replayed as it cannot be seen in clear text.
Brian West schrieb:
Its called TLS...
/b
On Oct 21, 2008, at 4:30 PM, Peter P GMX wrote:
can you post debug logs of this output?
Mike
On Oct 23, 2008, at 6:43 AM, Birgit Arkesteijn wrote:
Hi,
Thanks Anthony for your response.
Unfortunately removing the 'kick' didn't make a difference.
The result is also different if I change the order of the UUIDs
around.
In one scenario I
2008/10/22 Anthony Minessale [EMAIL PROTECTED]:
event socket has the command sendmsg which lets you send a message to a
specific channel. This can be any message but the one you are familiar with
is the one that tells the session to execute an application. Think of it as
you are sending an
Hi Mike,
The two snippets can be found below.
Thanks, Birgit
* The snippet where [DESTINATION_OUT_OF_ORDER] occurs:
2008-10-23 12:47:06 [NOTICE] sofia.c:2110 sofia_handle_sip_i_state()
Channel [sofia/external/0663] has been answered
2008-10-23 12:47:06 [DEBUG] switch_ivr_originate.c:1322
How much lower?
it's microsecond accuracy afterall.
There is just a timestamp set when a 180 or 183 is received.
its calculated by subtracting that time from the time the call origination
started.
On Thu, Oct 23, 2008 at 3:09 AM, shehzad p [EMAIL PROTECTED] wrote:
Hi again,
Now i found two
Here are some more videos from the last ClueCon
http://files.freeswitch.org/cluecon_2008/
On Thu, Oct 23, 2008 at 5:03 AM, Thomas Troesch [EMAIL PROTECTED]wrote:
Thank you for your input. I am clearly naive about what may be required,
and I need to get more details on what is currently in
Who are you calling with the gateway? As in what provider/SIP agent?
If caller-id-in-from makes it work but breaks something else that means who
you are calling expects it to be in the From: header
which is often done but has evolved into being incorrect these days.
We also send a Remote-Party-ID
On Thu, Oct 23, 2008 at 6:45 AM, Dennis [EMAIL PROTECTED] wrote:
2008/10/22 Anthony Minessale [EMAIL PROTECTED]:
event socket has the command sendmsg which lets you send a message to a
specific channel. This can be any message but the one you are familiar
with
is the one that tells the
See this one?
2008-10-23 14:38:49 [NOTICE] mod_spidermonkey.c:2860 session_destroy()
Hangup sofia/external/0662 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
try
session.setAutoHangup(false);
to keep the call from getting hungup when the script exits.
BTW, you seem to be using older code. Keep in mind
Thanks for the info. I was also missing that responses sent by Freeswitch are
also terminated with double CR.
Quick followup question: I noticed that on the event socket interface, for any
single DTMF digit pressed, Freeswitch sends two DTMF events. One with
Channel-State: CS_EXCHANGE_MEDIA
2008/10/23 Anthony Minessale [EMAIL PROTECTED]:
Please do not lose patience with me. I am absolutely new to the whole
sip-/phone-/call-thing...
What messages do you want to send? Be Specific.
For example I want to send the message playfile from one process to
another (the call_direction
What does 123 lead to? launching another script?
you could make the 2nd originate lead to park() in place of 123
and you could control both legs from the same socket.
please please join irc and let the whole group help you.
On Thu, Oct 23, 2008 at 10:02 AM, Dennis [EMAIL PROTECTED] wrote:
Hi Anthony,
Well, that definitely improved the situation, thanks!
I'm running 498:8901, installed on 2008-10-08.
The problem I now hit is that the two calls are bridged indeed, but the
first person (0662) remains in the conference as well. See lines below.
Since the first person was alone in
2008/10/23 Anthony Minessale [EMAIL PROTECTED]:
What does 123 lead to? launching another script?
The 123 is the destination_number in the /dialplan/default.xml.
extension name=test
condition field=destination_number expression=^123$
action application=set
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
ist there a way in freeswitch to set the screening indicatoron TDM side
like asterisk's SetCallerPres resp CallerPres does ?
regards
Helmut Kuper
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.9 (MingW32)
Comment: Using GnuPG with Mozilla
On Oct 23, 2008, at 11:23 AM, Birgit Arkesteijn wrote:
Hi Anthony,
Well, that definitely improved the situation, thanks!
I'm running 498:8901, installed on 2008-10-08.
revision 8901 is from july 7 of this year. I would say you would have
to update to trunk and try this again to be sure
Hi,
I'm running 'make current' as I type and will get back to you to report
if the problem is solved in version 596:10131 .
Thanks, Birgit
On 23/10/08 17:36, Michael Jerris wrote:
On Oct 23, 2008, at 11:23 AM, Birgit Arkesteijn wrote:
Hi Anthony,
Well, that definitely improved the
Hi,
Rebuild went quicker than I thought.
Unfortunately the problem is still there, i.e. two channels and one
conference call, where 0662 is in both.
[EMAIL PROTECTED] show channels
API CALL [show(channels)] output:
what are you doing again?
I think I lost track?
On Thu, Oct 23, 2008 at 12:09 PM, Birgit Arkesteijn
[EMAIL PROTECTED]wrote:
Hi,
Rebuild went quicker than I thought.
Unfortunately the problem is still there, i.e. two channels and one
conference call, where 0662 is in both.
[EMAIL
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