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
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
http://wiki.freeswitch.org/wik
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
___
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 t
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:
While SIP Session Timers parameters are those:
Am I right? Thanks a lot.
--
Iñaki Baz Castillo
__
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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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
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 t
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,
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.
> Should
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
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 billin
Hi,
Our customer has different requirements for the situation where a person
answered a call or a answerphone.
I found a solution on:
http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation
This solution requires a person to press a DTMF.
However, I'd much rather do it the
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
I have the corresponding cidr
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 tr
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 fo
Birgit,
I'm pretty sure that ringing is early media and you've got
ignore_early_media set to true. If you can work without ignoring early
media then you'll get rings. However, there's probably a reason you're
ignoring early media so be sure to test thoroughly without ignoring
early media j
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!
_
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!
__
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
Hi Baskar,
I'm not sure why you get the "-ERR no reply" answer since that answer is
never given by the function itself.
Nevertheless the uuid_send_dtmf function sends the dtmf_data TO the uuid you
enter, for what you want to achieve you need to "simulate" that the channel
that is eavesdropp
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
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 *
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
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 a
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 ch
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.
>
>
Now I need to see the console logs with debug set. "console loglevel
debug" or press F8, Also document the exact scenario and every step
you take, every key you press.
/b
On Nov 18, 2008, at 9:04 AM, x y wrote:
Datavox Ip-300
___
Freeswitch-us
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
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
Freeswitch-users@lis
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.
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo
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
These aren't inserting 1003 as the caller_id_number are they?
/b
On Nov 18, 2008, at 5:19 AM, Baskar wrote:
>
>
>
>
>
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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 inboun
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 * the playback is stopped and i
receive an channel_execute_complete playback for this file. this only
happens with the * - the other keys do nothing
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 mec
Hi,
I am working on Dtmf signal on barging
I have added these line to default.xml
i eavesdrop on extension 1003 by dialing 881003
1003 is connected to mobile no
but when i dial 881003 it get hangup
i hav
The rfc also describes why:
SIP provides a mechanism to represent common header field names in an
abbreviated form. This may be useful when messages would otherwise
become too large to be carried on the transport available to it
(exceeding the maximum transmission unit (MTU) when using U
2008/11/18 Saurabh Aggarwal <[EMAIL PROTECTED]>:
> enabling compact headers - what is that?
SIP allows compact headers names for a few heades:
From = f
To = t
Via = v
...
--
Iñaki Baz Castillo
<[EMAIL PROTECTED]>
___
Freeswitch-users mailing list
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
Its not really possible other then enabling compact headers or by getting
rid of codecs that you don¹t actually want to use... Another thing you could
do is get your broken ISP to fix their firewall... It is not correct to just
drop fragmented packets just because they are fragmented.. This is some
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 an
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 size
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