Ok, my bad. Ethereal for some reason was showing only the first fragment
(ethereal bug?).
But, now it seems I have hit another problem - it seems that the SIP invites
(which are fragmented) are being dropped by the firewall in between us and the
SIP provider. Is it possible to shrink the
2008/11/18 Saurabh Aggarwal [EMAIL PROTECTED]:
Ok, my bad. Ethereal for some reason was showing only the first fragment
(ethereal bug?).
But, now it seems I have hit another problem - it seems that the SIP invites
(which are fragmented) are being dropped by the firewall in between us and
the
enabling compact headers - what is that?
-Saurabh
Date: Tue, 18 Nov 2008 04:29:28 -0600From: [EMAIL PROTECTED]: [EMAIL
PROTECTED]: Re: [Freeswitch-users] SIP Invite IP fragmentation issueIts not
really possible other then enabling compact headers or by getting rid of codecs
that you don’t
Thanks, how do I enable this in freeswitch? Can this be done through the SIP
configuration file?
-Saurabh
Date: Tue, 18 Nov 2008 12:05:18 +0100From: [EMAIL PROTECTED]: [EMAIL
PROTECTED]: Re: [Freeswitch-users] SIP Invite IP fragmentation issueThe rfc
also describes why:SIP provides a
hi,
i would like to be able to listen to conversations, while they are
ongoing. this should not happen over a phone. i would like to be able
to have a link or something in my admin-area, where i can click, if i
want to listen to a conversation.
i thought about to start a record with socket
These aren't inserting 1003 as the caller_id_number are they?
/b
On Nov 18, 2008, at 5:19 AM, Baskar wrote:
action application=db data=insert/spymap/$
{caller_id_number}/${uuid}/
action application=db data=insert/last_dial/$
{caller_id_number}/${destination_number}/
Hy!
I have tested several times so far the att_xfer function of the freeswitch, and
I've found it unstable. I'm
using a similar code to the example at the freeswitch wiki, and sometimes a
call isnt transfered.
The att_xfer works fine in the cases as I figured out from the log, when
What device are you using?
/b
On Nov 18, 2008, at 8:03 AM, x y wrote:
Im using latest fs, and sip phones without built in transfer.
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Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
Datavox Ip-300, and X-Lite softphone for testing.
Hirdetés (x)
Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a
kötelező biztosítások kiindulópontja!
___
Freeswitch-users mailing list
Hi, I've read that FS supports/implements Session Timers to monitorice
both legs of a call. How to enable it? I mean:
alice --- FS bob
- alice calls bob vía FS
- FS calls bob.
- bob answers (sends 200 OK).
- bypass_media mode, no RTP through FS.
- FS establishes a SIP dialog with
Hi,
Thanks, Anthony, for your reply.
mSession.sleep(100);
does indeed do the trick!
Cheers, Birgit
On 17/11/08 19:51, Anthony Minessale wrote:
you never want to msleep during a js running on a call
you should use session.sleep(500);
msleep blocks the whole thread and thus the audio.
On
Hi,
In Javascript, I do the following:
var mSession = new
Session({ignore_early_media=true,originate_timeout=8} + endpoint_url);
mSession.execute(record_session, recordfile);
This works very well, however, the client would like the rings to be
recorded as well.
I've tried using various
you can use the eavesdrop dialplan app from a new call to spy on an in
progress session
it takes the uuid of the channel you want to listen to as the arg.
On Tue, Nov 18, 2008 at 6:28 AM, Dennis [EMAIL PROTECTED] wrote:
hi,
i would like to be able to listen to conversations, while they are
set the variable playback_terminators=none before you execute playback.
On Tue, Nov 18, 2008 at 6:21 AM, Dennis [EMAIL PROTECTED] wrote:
i am using socket outbound and if an inbound call comes in, i answer
the call and play a soundfile for the caller.
if the caller presses the dtmf key *
Devices: IP-300 with 1007 registered
IP-300 with 1009 registered
X-Lite with 1011 registered
Situation where att_xfer success:
-Dial IP-300 with 6691007 from X-Lite (669 is just a prefix, assuring to
execute my extension)
-1007 ringing, get the phone
-press
the only reliable answer is use TCP.
The RFC is daft in this matter.
They say when it's bigger than mtu to automatically use TCP instead.
And timeout for 10 seconds then fall back to UDP.
Its mandatory in SIP to support both TCP and UDP up to 64k per packet.
As you can see, since barely anything
You missed one thing. the console log with debug.
/b
On Nov 18, 2008, at 9:57 AM, x y wrote:
Devices: IP-300 with 1007 registered
IP-300 with 1009 registered
X-Lite with 1011 registered
Situation where att_xfer success:
-Dial IP-300 with 6691007 from X-Lite (669
And btw, sorry for my english... thinked and tought, in one
sentenceif only my english teacher would saw this :)
Hirdetés (x)
Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a
kötelező biztosítások kiindulópontja!
I tought u thinked of these:
http://pastebin.freeswitch.org/6179
http://pastebin.freeswitch.org/6178
If not, please correct me .
Viktor
Hirdetés (x)
Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a
kötelező biztosítások kiindulópontja!
Hi Michael,
Thanks for your reply.
I tried a variety of options:
var mSession = new Session(endpoint_url);
var mSession = new Session({ignore_early_media=false} + endpoint_url);
both only return after the session is answered (originated?).
The customer wants
the number of seconds we wait for
Birgit,
I'm almost to my office. I will give you more info soon. I have not
used js in this capacity so we will have to do some experimenting.
-MC
Sent from my iPhone
On Nov 18, 2008, at 9:51 AM, Birgit Arkesteijn [EMAIL PROTECTED]
wrote:
Hi Michael,
Thanks for your reply.
I tried a
I am using acls (cidr) to accept incoming calls from a gateway that
I do not want to register in my FS box.
I have this gateway configured in a xml file :
freeswitch/conf/directory/default/gateway1.xml
include
user id=GATEWAY1 mailbox= cidr=xxx.xxx.xxx.xxx/32
params
param
Add ${accuntcode} to the CDR template in cdr.conf.xml... the template
can include any variables from the session.
/b
On Nov 18, 2008, at 12:50 PM, [EMAIL PROTECTED] wrote:
Any help on how to define an endpoint (originating) and use some
attribute (like account_code or user id)
for billing
The ${accountcode} variable IS set in the cdr_csv.xml conf file yet the
field is empty after the call.
Shouldn't it also show in the xml cdr? I thought the XML CDRs included
all of the session variables.
Brian West wrote:
Add ${accuntcode} to the CDR template in cdr.conf.xml... the template
I'm going to guess that this is an inbound call to the user. Which
means the variables aren't set inbound to the user.
/b
On Nov 18, 2008, at 1:40 PM, [EMAIL PROTECTED] wrote:
The ${accountcode} variable IS set in the cdr_csv.xml conf file yet
the field is empty after the call.
Yup, I tried different settings for dft_min and dft_confirm before but did not
understand what they meant.
Back then I guessed that dft meant discrete Fourier transform not default
and these variables were cut off frequencies of a filter but that did not quite
make sense.
For instance, in
you have to manually set the var on the channel in your dialplan.
On Tue, Nov 18, 2008 at 1:40 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
The ${accountcode} variable IS set in the cdr_csv.xml conf file yet the
field is empty after the call.
Shouldn't it also show in the xml cdr? I
El Martes, 18 de Noviembre de 2008, Iñaki Baz Castillo escribió:
Hi, I've read that FS supports/implements Session Timers to monitorice
both legs of a call. How to enable it? I mean:
alice --- FS bob
- alice calls bob vía FS
- FS calls bob.
- bob answers (sends 200 OK).
-
yes but RTP timers are in there too.
/b
On Nov 18, 2008, at 5:26 PM, Iñaki Baz Castillo wrote:
They seem to be related to SIP Session Timers (nothing related to
RTP), am I
right?
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Freeswitch-users mailing list
El Miércoles, 19 de Noviembre de 2008, Brian West escribió:
yes but RTP timers are in there too.
Well, but I expect that RTP timers parameters are the following:
param name=use-rtp-timer value=true/
param name=rtp-timer-name value=soft/
param name=rtp-timeout-sec value=300/
param
Hi All,
We all like to be thanked and when someone does something for me, I like to
show my gratitude towards them and I think what Anthony did for me today
deserves a public show of gratitude. It goes like this:
I have a single port FXO card in my home machine running FS and I needed to
modify
Dear all,
Hi
I ve just started reading on freeswitch. From where should i
start? Basically, i am am a graduate electronic engineer. Can i be a successful
developer of freeswitch? Its my first experience of any telephonic system.
Faisal rehman, Pakistan
On Wed, Nov 19, 2008 at 11:07 AM, Faisal Maqsoodi
[EMAIL PROTECTED]wrote:
Faisal:
Welcome to the wonderful word of open source telephony.
I ve just started reading on freeswitch. From where should i
start?
http://wiki.freeswitch.org/wiki/Main_Page
Thanks very much sir u r so nice n cooperative.
--- On Tue, 11/18/08, Wasim Baig [EMAIL PROTECTED] wrote:
From: Wasim Baig [EMAIL PROTECTED]
Subject: Re: [Freeswitch-users] i need help regarding freeswitch
To: freeswitch-users@lists.freeswitch.org
Date: Tuesday, November 18, 2008, 10:36 PM
On
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