Zaptel isn't required for sangoma. Just make sure when installing
wanpipe the value you select is the TDM API with the words FreeSWITCH
in the menu option.
/b
On Sep 19, 2008, at 12:57 AM, Gopal krishnan wrote:
Hi Anthony,
My conf as follows,
openzap.conf
[span wanpipe]
trunk_type =
Hello,
is there a way to interconnect fs to a Cisco Call Manager which is
configured to speak SCCP protocol (aka Skinny) and not SIP?
I did not found a mod_SCCP in the docs :-)
Thanks,
Claudio
Internet Email Confidentiality Footer
There is no SCCP module for FS. CM only uses SCCP to talk to phones, it
uses either MGCP or SIP to talk to gateways. So if you have a version
that has SIP support (I believe 4.0), then you could connect CM to FS.
Cavalera Claudio Luigi wrote:
Hello,
is there a way to interconnect fs to a
[EMAIL PROTECTED] wrote:
There is no SCCP module for FS. CM only uses SCCP to talk to phones,
it uses either MGCP or SIP to talk to gateways. So if you have a
version that has SIP support (I believe 4.0), then you could
connect CM to FS.
It would be possible in this way to assign to ring
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
On Fri, Sep 19, 2008 at 6:41 PM, Gopal krishnan [EMAIL PROTECTED] wrote:
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
Hello,
mod_sofia don't start correctly when there isn't a internet connection.
We can use fs only in lan (without internet connection)? Is there a
particolar configuration for this situation?
Regards,
--
Rocco Lucente
Vi preghiamo di considerare l'ambiente prima di stampare questa e-mail
On Sep 19, 2008, at 5:48 AM, Rocco Lucente wrote:
Hello,
mod_sofia don't start correctly when there isn't a internet
connection.
We can use fs only in lan (without internet connection)? Is there a
particolar configuration for this situation?
Regards,
The default configuration includes
I have tested the option of adding channel variables to the bridge
string, and it does not work.
This dialplan works:
extension name=External calls
condition field=destination_number expression=^(\d{8})$
action application=set data=effective_caller_id_number=45161061/
action application=bridge
Hi,
Since I am not able to make the outbound call, when I use this command oz
dump 1 a 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:
On Sep 19, 2008, at 9:07 AM, Jon Bruel wrote:
I have tested the option of adding channel variables to the bridge
string, and it does not work.
This dialplan works:
extension name=External calls
condition field=destination_number expression=^(\d{8})$
action application=set
On Sep 19, 2008, at 9:18 AM, Gopal krishnan wrote:
Hi,
Since I am not able to make the outbound call, when I use this
command oz dump 1 a 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:
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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Also if the far side is Asterisk and they don't have trustrpid=yes
then you have to set the param on the gatway:
param name=caller-id-in-from value=true/
/b
On Sep 19, 2008, at 8:20 AM, Michael Jerris wrote:
effective_caller_id_number is meant to be set on the a leg, not the b
leg.
Mike
IDLE would make more sense.
/b
On Sep 19, 2008, at 8:22 AM, Michael Jerris wrote:
in openzap, state down is like on-hook... we should change that name
maybe.
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oops...sorry for the mess...
import org.apache.mina.common.IoSession
import java.util.concurrent.BlockingQueue
import java.util.concurrent.Executors
import java.util.concurrent.ExecutorService
import java.util.concurrent.Future
import java.util.concurrent.Callable
import
Mike, you answered: effective_caller_id_number is meant to be set on the
a leg, not the b-leg.
Well I need to control the effective_caller_id_number (or whatever it
is called in the B-leg) individually for each location when I bridge to
many destinations. In my case, I need to have different
On Fri, Sep 19, 2008 at 9:02 AM, Jon Bruel [EMAIL PROTECTED] wrote:
Mike, you answered: effective_caller_id_number is meant to be set on the
a leg, not the b-leg.
Well I need to control the effective_caller_id_number (or whatever it
is called in the B-leg) individually for each location when
no it wouldn't
When the channel is not in use it's down.
it's absent of any activity and not running in the state machine.
This terminology is for the sake of the coder not the guy using the code.
On Fri, Sep 19, 2008 at 8:40 AM, Brian West [EMAIL PROTECTED] wrote:
IDLE would make more
the FS site says FS supports TLS etc so wouldnt it be good if the windows
binary were compiled with TLS as by default they r not so guys like me who
actually download the msi and install and run it can atleast have TLS
support by default on windows platform rather than them trying to make it
work
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
On Sep 19, 2008, at 10:46 AM, xbipin wrote:
the FS site says FS supports TLS etc so wouldnt it be good if the
windows
binary were compiled with TLS as by default they r not so guys like
me who
actually download the msi and install and run it can atleast have TLS
support by default on
Use the pastebin please.
On Sep 19, 2008, at 7:00 AM, Richard Open Source [EMAIL PROTECTED]
wrote:
oops...sorry for the mess...
import org.apache.mina.common.IoSession
import java.util.concurrent.BlockingQueue
import java.util.concurrent.Executors
import
Large pastes on the mailing list are ok... if they aren't over 100k :P
/b
On Sep 19, 2008, at 10:24 AM, Christian Jensen wrote:
Use the pastebin please.
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None of the suggestions worked, and I still can't control the A-number
individually when bridging to multiple destinations.
I have tried to change the dialplan using
[origination_caller_id_number=1234]
As a part of the string in the bridge data, but an info after bridge did
not show the
Is the far side Asterisk? If so then I suspect they aren't trusting
the RPID you're sending. They'll have to trustrpid.
/b
On Sep 19, 2008, at 10:56 AM, Jon Bruel wrote:
None of the suggestions worked, and I still can't control the A-number
individually when bridging to multiple
On Thu, Sep 18, 2008 at 3:49 PM, Luke Graybill [EMAIL PROTECTED] wrote:
My suggested solution is to apply the job-id concept from bgapi to messages
as well, and to go a step further; borrow the Asterisk idea of transmitting
an identifier along with each command. Every response and event
On Sep 19, 2008, at 6:23 AM, Gopal krishnan wrote:
Hi,
Basically I just want to test outbound alone with freeswitch, so I
can use extensions.conf in the conf directory rite?
--
I would forget about the asterisk dialplan then.
It's very simple to configure an outbound SIP provider in
Oops, I forgot to mention anthm as well - he provided great feedback on irc!
On Fri, Sep 19, 2008 at 10:19 AM, Luke Graybill [EMAIL PROTECTED] wrote:
On Thu, Sep 18, 2008 at 3:49 PM, Luke Graybill [EMAIL PROTECTED] wrote:
My suggested solution is to apply the job-id concept from bgapi to
On Sep 19, 2008, at 12:19 PM, Luke Graybill wrote:
On Thu, Sep 18, 2008 at 3:49 PM, Luke Graybill [EMAIL PROTECTED]
wrote:
My suggested solution is to apply the job-id concept from bgapi to
messages as well, and to go a step further; borrow the Asterisk idea
of transmitting an
We don't see much samples for PRI outbound dialing . India the line is Euro
ISDN.
Has anyone tested sangoma A101 cards with Openzap ?
I am tring to build a front end web application , to dial using JS in FS
which will dial a outbound no and bridge the call to the extension.
Thank you
Imthiyaz
Hi,
I'm trying to figure out how to use the command API via XML-RPC. I want to do
the two following things one after the other: 1) place a call, and 2) play an
audio on the newly created call.
The Java code for placing a call (copied from wiki) works fine and is following:
In this case you wouldn't use XML RPC you would use mod_event_socket.
http://wiki.freeswitch.org/wiki/Mod_event_socket#SendMsg
telnet localhost 8021
auth ClueCon enter enter
sendmsg 32a5e11a-8649-11dd-bb78-fd02030a93ef
call-command: execute
execute-app-name: playback
execute-app-arg:
As the author of ASTPP I'd be very interested in negotiating a bounty to
speed things up. User authentication is working with Freeswitch as well
as lcr and call rating. It's largely a matter of documenting and testing.
Darren Wiebe
[EMAIL PROTECTED]
xbipin wrote:
are u interested in a paid
There should be nothing major keeping it from working on windows. It's
based on perl and mysql. There are ports of both perl and mysql for
windows.
Darren Wiebe
[EMAIL PROTECTED]
Michael Jerris wrote:
On Sep 18, 2008, at 12:00 PM, xbipin wrote:
are u interested in a paid
I haven't encountered every event yet (I'm still in the process of writing
my client) but perhaps some others can help? I'll post the ones I know about
so far though..
- command/reply
- CHANNEL_EXECUTE
- CHANNEL_EXECUTE_COMPLETE
- CHANNEL_ANSWER
- CHANNEL_PARK
The last two are uncertain because,
No the far side is BroadWorks, and as I said in the first mail, the only
difference between an accepted and a rejected INVITE is in the contents
of the Remote-Party-ID header. Generally in many European countries, the
operators limit the A-number to the actual number of the line dialled
out on.
Can you show me a side by side comparison of working vs non-working?
/b
On Sep 19, 2008, at 12:37 PM, Jon Bruel wrote:
No the far side is BroadWorks, and as I said in the first mail, the
only
difference between an accepted and a rejected INVITE is in the
contents
of the Remote-Party-ID
I've pretty much standardized on using g729 as the codec of choice for
the users connecting to my FS installation, primarily due to the users I
have that are on very slow connections and all of my carriers support
g729, so no need to transcode.
The sounds files that comes with FS are raw PCM
whats the amount ur looking for this bounty, mayb guys like me who wanna use
it as a proper softswitch can pitch in a little and can also help in testing
and coding.
Darren Wiebe wrote:
As the author of ASTPP I'd be very interested in negotiating a bounty to
speed things up. User
On Sat, Sep 20, 2008 at 2:07 AM, xbipin [EMAIL PROTECTED] wrote:
whats the amount ur looking for this bounty, mayb guys like me who wanna
use
dwiebe can let you know more about that
it as a proper softswitch can pitch in a little and can also help in testing
and coding.
what is sorely
Yes I can, but somehow I start to repeat myself. An INVITE is sent
allright every time, but one of the headers differs:
Works:
Remote-Party-ID:
sip:[EMAIL PROTECTED];screen=yes;privacy=off
Doesn't work:
Remote-Party-ID: Extension 1000
sip:[EMAIL PROTECTED];screen=yes;privacy=off
I have tried with
Let me show you some examples that I'm doing right now that work fine.
I do this at the CLI and it works:
originate {origination_caller_id_number=9184238080}sofia/gateway/
asterlink.com/1918XXX
And this is in my dialplan works:
extension name=asterlink.com
condition
Hello -
Got an issue with Freeswitch not responding on the port that the
initial request was made on. I'm not beyond believing that it is a NAT
or router issue except that I can register a Cisco phone from another
location or a softphone from the same location without any problem.
This
On Sep 19, 2008, at 6:08 PM, David Aldworth wrote:
Hello -
Got an issue with Freeswitch not responding on the port that the
initial request was made on. I'm not beyond believing that it is a
NAT or router issue except that I can register a Cisco phone from
another location or a
Also what is version and make sure you're on svn trunk if possible.
That [] had a bug in it at some point and I can't recall the exact rev
we fixed it in. But I'm on svn trunk here working to nail down bugs
so we can release 1.0.2.
/b
___
post a trace of FS after pressing f8 from the cli
detailing the entire call and we can have a look.
On Fri, Sep 19, 2008 at 6:29 PM, Matt Darnell [EMAIL PROTECTED] wrote:
On Wed, Sep 17, 2008 at 10:14 AM, Matt Darnell [EMAIL PROTECTED]
wrote:
On Fri, Jul 25, 2008 at 8:51 PM, UV [EMAIL
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