Hi Fred,
Yes you can use Sangoma USB FXO with your laptop. You need to install
openzap for this. But for testing you can use this driver. Still there is
some issue with Openzap with FS as for as I used. while installing Sangoma
USB FXO device you need to use beta drivers.
On Sun, Mar 1, 2009 at
Hi,
Actually what is the difference between ESL in FS 1.0.3 and event socket
in FS 1.0.2. Is the FS 1.0.3 ESL superior?
On Fri, Feb 27, 2009 at 6:43 PM, Rex_Alex wrote:
> Hi All, I did what you have all suggested. Now its working perfectly.
> Thanks a lot for all your assistance. Rex.
>
> Raym
Hi Brian,
Please find the attached backtrace files attached.
And
1. SVN revision number (or binary file) - FreeSWITCH Version 1.0.3
(exported)
2. Operating System and revision - CentOS 5.2
3. Hardware information - 32 bit with 512 MB RAM
4. I am using Event socket
5. Language - Javascript
One mo
Hi,
I tried like this
api uuid_getvar endpoint_disposition but when the far end is ringing
its showing as ANSWER in the telnet console since my local extension is
answered.
I would like to show the endpoint_disposition for the far end.
my originate command is
api originate sofia/interna
Hi,
Can this would help us
uuid_getvar
like *api uuid_getvar Answer state:*
**
--
Thank you with regards,
Gopal,
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Hi
I need to get the channel answer state for particular user. For example
there is a GUI for outbound call where the a ajax program will run in the
background of the program to get the answer state. after i originate a call
from event socket the ajax program will start monitor the line answer
Hi,
Thanks for the mail. I tried in this format to detect the busy signal but
I cant.
I am using javascript file like,
session1 = new Session();
session1.originate(session1,"{ignore_early_media=true}sofia/internal/"+argv[0]+",30");
session1.execute("bridge", "sofia/default/"+argv[1]+"@172.20.176
Hi,
I am using event socket to originate calls. I need to originate the
calls thru console and need to detect the tone. In Asterisk we used to
detect thru BackgroundDetect and VMDetect. In freeswitch I found that the
tones.conf which will detect the tones that we are dialing. I am not sure
how
Hi,
Auto responder you need to create by an IVR application. IVR is a pre
recorded message with some dtmf signalling by which once the inbound call
comes in a welcome message will be played and the IVR will prompt the user
to enter the keys thru dtmf based on the key entered by the user the IV
Hi,
Is it possible to have busy tone detection thru console dialing?
--
Thank you with regards,
Gopal,
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UNSUBSCRIBE:
mand now I can able to get the channel
status for my outbound number.
Thanks for the help and IRC
On Mon, Oct 13, 2008 at 2:16 PM, Gopal krishnan <[EMAIL PROTECTED]> wrote:
> Hi,
> Let me explain bit more clear,
>
> I was dialing thru event socket in telnet. The events com
answers the events are not
getting parsed inside the telnet.
If I am doing anything wrong?
On Mon, Oct 13, 2008 at 12:55 PM, Gopal krishnan <[EMAIL PROTECTED]> wrote:
> Hi,
> I can able to see the events log as I disussed in IRC thru events plain
> CHANNEL_CREATE CHANNEL_ANSWER
or hangup.
Any help would be appreciated. Thanks
On Mon, Oct 13, 2008 at 11:26 AM, Gopal krishnan <[EMAIL PROTECTED]> wrote:
> Hi,
> Is it possible to get the answer state like the below by using uuid
>
> Answer state : ringing
>
> Answer state: answered
>
> A
> 2008/10/5 Michael S Collins <[EMAIL PROTECTED]>
>
> I don't know PHP. If no one else here does either then you'll need to ask
>> this question on a PHP list or IRC channel.
>>
>> -MC
>>
>> Sent from my iPhone
>>
>> On Oct 4, 2008, a
Hi,
We can use the recording in api as the one like the below,
api uuid_record start/path to record the file.
Thanks
On Thu, Oct 9, 2008 at 5:37 PM, Michael Jerris <[EMAIL PROTECTED]> wrote:
>
> On Oct 9, 2008, at 8:01 AM, Gopal krishnan wrote:
>
> > Hi,
> >
&
Hi,
I am trying to record thru telnet with sendevent record and also tried
sendevent record_session but I cant able to record. Is there any command to
record thru telnet?
--
Thank you with regards,
Gopal,
___
Freeswitch-users mailing list
Freeswitch
just the one your
> specify.
>
> /b
>
>
> On Oct 6, 2008, at 8:59 AM, Gopal krishnan wrote:
>
> > Hi,
> >
> > we tried this command and working fine for particular extension
> > hangup
> >
&
Hi,
we tried this command and working fine for particular extension hangup
fsctl hupall normal_clearing dialed_ext 1000 where 1000 is the extension
number.
On Mon, Oct 6, 2008 at 4:30 PM, Gopal krishnan <[EMAIL PROTECTED]> wrote:
> Hi,
> I am trying to hangup a bridged ca
Hi,
I am trying to hangup a bridged call via event socket, but the channel is
not getting hangup. I tried with the following commands in telnet,
sendevent channel_hangup - no output
sendevent hangup - no output
sendmsg
any other manager commands are there to hangup a channel?
--
Thank you
the preg
> functions in your PHP version? If not then you'll need it for this to work.
>
> -MC
>
>
> --
>
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *Gopal
> krishnan
> *Sent:* Friday, Octob
t data reliably and acting accordingly.
>
>
> --
>
> *From:* [EMAIL PROTECTED] [mailto:
> [EMAIL PROTECTED] *On Behalf Of *Gopal
> krishnan
> *Sent:* Friday, October 03, 2008 10:33 AM
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* Re:
t; Yes if you parse the event using something like perl, ruby, php and get
> it...
> /b
>
> On Oct 3, 2008, at 12:10 PM, Gopal krishnan wrote:
>
> File attached
>
> On Fri, Oct 3, 2008 at 10:36 PM, Gopal krishnan <[EMAIL PROTECTED]> wrote:
>
>> Hi,
>> B
File attached
On Fri, Oct 3, 2008 at 10:36 PM, Gopal krishnan <[EMAIL PROTECTED]> wrote:
> Hi,
> By giving event channel_answer in telnet console I get lots of variables,
> I am attaching it as a text file with this email. And my query is for
> example If I want to pickup
AIL PROTECTED]> wrote:
> like i said, do events all
> then watch them all on telnet for a sample call and decide for yourself
> which ones you need.
> there is a CHANNEL_ANSWER for instance that you might find interesting ;)
>
>
>
> On Fri, Oct 3, 2008 at 2:15 AM, Gopal kris
t; events channel_state channel_destroy
>
>
>
>
>
>
> On Thu, Oct 2, 2008 at 8:38 AM, Gopal krishnan <[EMAIL PROTECTED]> wrote:
>
>> Hi,
>>
>>I am trying to execute sendevent channel_state in a php program via
>> socket, but I didn'
annel's current status.
>>> You could later reference those variables directly in your xml / script
>>> dialplan.
>>>
>>>
>>> -- Forwarded message --
>>>> From: "Gopal krishnan" <[EMAIL PROTECTED]>
>>&g
and after
> bridge, you will see a lot of variables with the channel's current status.
> You could later reference those variables directly in your xml / script
> dialplan.
>
>
> -- Forwarded message ------
>> From: "Gopal krishnan" <[EMAIL PROTECTED]>
Hi,
Is there any possibilities that I can check my channel status whether it
is ringing or answer or hangup. I am trying to fetch thru uuid but couldn't
able to do that.
any suggestion would be helpful. thanks
--
Thank you with regards,
Gopal,
___
u can have a large
> number of group combos
> (it's not limited to 3 digits)
>
> 80XXX remove yourself from group XXX
> 81XXX add yourself to group XXX
> 82XXX call everyone in group XXX at the same time
> 83XXX call everyone in group XXX one at a time until someone answers
Hi,
Is there a possible way as like in Asterisk where the agents will login in
queue, so that the established call will be directly transferred to the
extension. Is there any module for that?
Any help would be appreciated. Thanks
--
Thank you with regards,
Gopal,
_
Hi,
Outbound call went thru socket with the PHP program. The program as follows,
";
echo "Caller is: $Caller";
echo "Callee is: $Callee";
$handle = fsockopen('127.0.0.1', '8021', $erno, $errstr, 30);
if (!$handle) {
//if connection fails exit and tell us what went wrong.
die("Connect failed wi
F4D27B19A9E3ACB1BF65D%0D%0A&variable_remote_media_ip=
172.20.179.201
&variable_remote_media_port=29188&variable_read_codec=PCMU&variable_read_rate=8000&variable_write_codec=PCMU&variable_write_rate=8000&variable_endpoint_disposition=RECEIVED]
So here I the key value and th
Hi,
I am trying to have a webpage with a button, once clicked the dial function
will happen, for this I hope we need to use mod_xml_curl. am i correct?
I have gone thru the link for the configuration of mod_xml_curl
http://wiki.freeswitch.org/wiki/Mod_xml_curl
I want to use javascript or poss
; Original Message:
> -----
> From: Gopal krishnan [EMAIL PROTECTED]
> Date: Tue, 23 Sep 2008 19:31:07 +0530
> To: freeswitch-users@lists.freeswitch.org
> Subject: [Freeswitch-users] Freeswitch with Audiocode Mediant 2000
>
>
> Hi,
>
> I followed the below l
Hi,
I followed the below link to configure the Audiocode Mediant 2000 with
Freeswitch
http://wiki.freeswitch.org/index.php?title=Configuring_AudioCodes_MP-114/118&printable=yes
but the above link is for FXO line, where I am using digital PRI line.
when I try to dial I am getting call failed,
Hi,
Any suggestion on this? Thanks
On Sat, Sep 20, 2008 at 6:50 PM, Gopal krishnan <[EMAIL PROTECTED]> wrote:
> Hi,
>
> My outbound is working, but not in a regular functionality, If i call
> first time the call goes thru, by second time its not getting thru, showing
>
Hi,
My outbound is working, but not in a regular functionality, If i call
first time the call goes thru, by second time its not getting thru, showing
that the channel is not getting released. After reloading mod_openzap the
channel becomes in DOWN state and the call is getting thru, so each an
Thanks for the reply,
I tried dialing a number and the pastebin link as follows,
http://pastebin.freeswitch.org/5611
and also I found that after dialed I saw the oz dump 1 2
and I found that the state is dialing and after few seconds automatically it
seems to hangup.
oz dump 1 3
API CALL [oz(
Hi Mike,
I changed the name in openzap.conf and also in default.xml, but the same
thing persisting, this hangup I terminated from the softphone, its not like
coming from the sip phone automatically. so Is there anything else I need to
check?
--
Thank you with regards,
Gopal,
__
Hi,
Basically I just want to test outbound alone with freeswitch, so I can use
extensions.conf in the conf directory rite?
--
Thank you with regards,
Gopal,
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Hi,
Since I am not able to make the outbound call, when I use this command in the console, I used to get the all the 31 channels , for a
refrence I am posing one channel block here,
*
span_id: 1
chan_id: 31
physical_span_id: 1
physical_chan_id: 31
type: B
state: DOWN
last_state: DOWN
cid_date:
Hi,
Is there any way that Asterisk dialplan can be used for freeswitch, since
there is a flle extensions.conf in PREFIX/conf directory, is th possible
with this file I can write a normal asterisk dialplan so that it will hit
the freeswitch. If possible how can it be done any examples or any wiki
this is an issue with openzap now, can you
>> clarify your openzap configuration and what kind of line it is hooked too
>> please?
>>
>> Mike
>>
>>
>> On Sep 18, 2008, at 3:55 AM, Gopal krishnan wrote:
>>
ore_session_run() OpenZAP/1:1/894929942 Running State Change
CS_CONSUME_MEDIA
so still the outbound is not yet thru. Thanks
On Wed, Sep 17, 2008 at 8:36 PM, Brian West <[EMAIL PROTECTED]> wrote:
>
> On Sep 17, 2008, at 7:24 AM, Gopal krishnan wrote:
>
> Hi,
>
> I am
Hi,
I am using Freeswitch with Sangoma A102 and Openzap. I have configured the
extension in default.xml as
*default.xml*
*
openzap.conf*
[span wanpipe]
trunk_type => e1
b-channel => 1:1-15
d-channel=> 1:16
b-channel => 1:17-31
*openzap.conf.xml*
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