It's normal to have to two records for a call - Start and Stop message.
From what i see - you have one start and stop for each leg of the call.
Regards,
AK
email lists wrote:
Hello,
Using the mod_radius_cdr module w/ Freeswitch, I am seeing duplicate
RADIUS messages being generated
*I want to move ILoadNotificationPlugin from being this “catch all” thing
that controls the entire assembly to something that can be used to fire up
code; effectively “OnLoad” and “OnUnload”. To dynamically control loading,
we’ll probably reflect on the individual plugins looking for attributes or
Hello *,
sched_api ...
works, too.
Thx again and looking forward to the next bug :)
Beni.
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how can i subscribe to custom event in cli.
cli: load mod_event_socket
say Module mod_event_socket Already Loaded!
but i use
cli: event plain CHANNEL_CREATE
return event: Command not found!
cli: api event plain CHANNEL_CREATE
return api: Command not found!
then what is the correct command?
the os have three ip, one public ipv4 with adsl which is dynamic assigned
every time, two lan ip in diffrent scope, 192.169.0.2 ,192.168.5.2.
when freeswitch start ,it auto bind to the pubic ip, so the lan user cann't
connect to freeswitch use lan ip.
i have setting
X-PRE-PROCESS cmd=set
jun yang yj13535428...@gmail.com wrote:
when freeswitch start ,it auto bind to the pubic ip, so the lan user cann't
connect to freeswitch use lan ip.
i have setting
X-PRE-PROCESS cmd=set data=bind_server_ip=0.0.0.0/
but have no effect, freeswitch also auto bind to the public ip.
any help is
i add
X-PRE-PROCESS cmd=set data=local_ip_v4=0.0.0.0/
before
X-PRE-PROCESS cmd=set data=domain=$${local_ip_v4}/
and it has no effect all the same.
is that something wrong.
2009/9/11 Jason White ja...@jasonjgw.net
jun yang yj13535428...@gmail.com wrote:
when freeswitch start ,it auto bind to
when i set local_ip_v4 to 0.0.0.0 i see the info below:
2009-09-11 20:22:27.15625 [WARNING] sofia.c:2291 Invalid IP 0.0.0.0 replaced
with 218.21.105.133
2009-09-11 20:22:27.15625 [WARNING] sofia.c:2300 Invalid IP 0.0.0.0 replaced
with 218.21.105.133
2009-09-11 20:22:27.15625 [NOTICE] sofia.c:1509
i also found that:
2009/7/17 Raul Fragoso raul at etellicom.com
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users:
* You can not do that with a single profile. Each profile is bound to only
** one local IP, so if you need to bind to more than one you will have to
** create a new
Hi,
I am having a strange problem here. sofia status shows that the user is
registered, but sofia_contact says the user is not registered.
Does anyone know why this is happening?
freeswi...@localhost.localdomain sofia status profile internal reg 180004
API CALL [sofia(status profile internal
Next Bug? Huh? :P
/b
On Sep 11, 2009, at 2:32 AM, Benedikt Fraunhofer wrote:
Thx again and looking forward to the next bug :)
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You need to telnet to the socket or use fs_cli... example...
telnet 0 8021
auth ClueConenterenter
events all plainenterenter (or what ever commands you wish to run)
/b
On Sep 11, 2009, at 3:48 AM, jun yang wrote:
how can i subscribe to custom event in cli.
cli: load mod_event_socket
say
You can NOT bind to 0.0.0.0 you can however use ${local_ip_v4} and if
the IP changes sofia will bounce the profile and update the IP.
/b
On Sep 11, 2009, at 7:55 AM, jun yang wrote:
i also found that:
2009/7/17 Raul Fragoso raul at etellicom.com:
You can not do that with a single profile.
No you should never be doing your billing inline like this. You
should be doing this externally of your application not inside your
dialplan.
/b
On Sep 10, 2009, at 11:40 PM, Ahmed Munir wrote:
Thanks for reply, well actually I'm doing billing after call hangup.
If h extension is
Thats normal too.
/b
On Sep 11, 2009, at 2:26 AM, Anatoliy Kounitskiy wrote:
It's normal to have to two records for a call - Start and Stop
message.
From what i see - you have one start and stop for each leg of the
call.
Regards,
AK
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http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_11
Here is the agenda please review and add to it anything you think we
should cover.
Thanks,
Brian
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FreeSWITCH is driven by a state machine and execute and hangup are opposing
states so once you change to hangup state that is the end of executing
extensions.
asterisk has 4 special extensions s h i and t we don't support any of them
because our dialplan concept and paradigm is completely
sip in general cannot properly support binding to 0.0.0.0 for a UAS, there
is no easy way for the sip stack to know which traffic is for which host and
all of the outbound traffic will appear to go out a single interface when no
specific binding is made.
running each ip on it's own profile is the
set the variable process_cdr=false on that a_leg first thing in your
dialplan
On Fri, Sep 11, 2009 at 2:26 AM, Anatoliy Kounitskiy
anato...@kounitskiy.com wrote:
It's normal to have to two records for a call - Start and Stop message.
From what i see - you have one start and stop for each
On Fri, Sep 11, 2009 at 4:01 PM, Brian West br...@freeswitch.org wrote:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_11
Here is the agenda please review and add to it anything you think we
should cover.
This time too, you all can follow the conference calling Skype the
skypeuser
You could create a daemon like this that listens for the
CHANNEL_HANGUP_COMPLETE event and send your CDR to the db.
http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/diegoviola/ruby/callcard/cdr.rb
Then do the billing stuff outside FreeSWITCH or use mod_nibblebill.
I suggest also that you
FYI, the conference is starting. Please join us!
sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org
213-799-1400
-MC
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On Fri, Sep 11, 2009 at 6:01 PM, Michael Collins m...@freeswitch.org wrote:
FYI, the conference is starting. Please join us!
sip:8...@conference.freeswitch.org
213-799-1400
This time too, you all can follow the conference calling Skype the
skypeuser skypiax5, then press 1 on the Skype dialpad
Dear Sir,
Is posible to play music for background when call connect ?
Example when i call my wife some time i need romantic song :)
BG
Dome C.
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if you want you could contribute a patch to make that a config option
(of course defaulting to the current value).
Mike
On Sep 4, 2009, at 5:51 AM, Peter P GMX wrote:
Thanks Anthony,
that did the trick.
Best regards
Peter
Anthony Minessale schrieb:
you can edit mod_xml_curl.c line 64
What errors do you get?
Mike
On Sep 4, 2009, at 8:35 AM, Muhammad Shahzad wrote:
Hi,
i am have FS SVN revision 14760, i am trying to use mod_xml_curl
against mod_dingaling. When i call xml_curl url in browser i get
mod_dingaling configuration correctly, also when i do reload
You can do it in perl or lua using a startup script that creates an
event listener.
Mike
On Sep 4, 2009, at 10:32 AM, Mathieu Parent wrote:
Hi
On Thu, Sep 3, 2009 at 6:52 AM, Mathieu Renemrene_li...@avgs.ca
wrote:
See
generally it keeps the overhead of running the script around during
the whole call.
Mike
On Sep 4, 2009, at 10:37 AM, Shameem Shiek wrote:
Hi Michael,
Why is it not recommended to do the brdge app right in the script?
The reason I ask this, I did have lot of trouble using Park/Fifo app
Please open a bug on http://jira.freeswitch.org for this issue.
Please test it on current svn trunk first as well.
Mike
On Sep 4, 2009, at 7:54 PM, DJB wrote:
I have a call transfer problem with Freeswitch
Here is the call flow:
I call from the PSTN (A party) into my Polycom phone
Following up, did a bug get created for this issue?
Mike
On Sep 5, 2009, at 2:40 AM, mayamatakeshi wrote:
On Fri, Sep 4, 2009 at 7:28 PM, mayamatakeshi
mayamatake...@gmail.com wrote:
On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi
mayamatake...@gmail.com wrote:
Hello,
I'm testing FS
actually, mod_dingaling is not reading configuration from xml_curl unless we
reload mod_dingaling, which obviously fails if dingaling profile is in call
etc.
So, i am writing a patch right now to enable this functionality, almost
finished just to perfect some memory management things.
Thank you.
I am trying to configure a Grandstream gateway to work with FS. I can make
outbound calls without a problem. However, inbound calls are getting a 403
Forbidden from FS in response to the INVITE from the gateway.
Now, the INVITE's from address is the caller's number (e.g. 111222),
which
Check out the variables ringback and transfer_ringback. The local extension in
the default dialplan is a good example.
For romance, I recommend 80s rock ballads. YMMV.
On September 11, 2009 12:22:20 pm Dome Charoenyost wrote:
Dear Sir,
Is posible to play music for background when
2009/9/12 Chris Burns ch...@cloudtel.com:
Check out the variables ringback and transfer_ringback. The local extension in
the default dialplan is a good example.
Music rinback is Ok now. but I'm looking for solution for stream sound
to channel both leg when call is answer.
For romance, I
Also for tests make sure you fuzz test it also .. giving it invalid
data shouldn't crash ... so try that when you're done too.
/b
On Sep 11, 2009, at 12:18 PM, Muhammad Shahzad wrote:
actually, mod_dingaling is not reading configuration from xml_curl
unless we reload mod_dingaling, which
On Fri, Sep 11, 2009 at 10:25 AM, Jerry Richards jerry.richa...@teotech.com
wrote:
I am trying to configure a Grandstream gateway to work with FS. I can make
outbound calls without a problem. However, inbound calls are getting a 403
Forbidden from FS in response to the INVITE from the
By the way, the FS DEBUG console is saying the following when an inbound
call is made:
Rejected by acl domains. Falling back to Digest auth.
Best Regards,
Jerry
-Original Message-
From: Jerry Richards [mailto:jerry.richa...@teotech.com]
Sent: Friday, September 11, 2009 10:25 AM
To:
When I try to d a load mod_sofia, I get an error message indicating that
Freeswitch can't find sofia.conf. There _is_ a sofia.conf.xml in the
autoload directory, which I _thought_ was the main sofia configuration
file. Do I need to copy it to sofia.conf? If so, where do I copy it to?
There are a few ways you could go about dropping into a conference and playing
the song in from a separate channel.
On September 11, 2009 01:47:43 pm Dome Charoenyost wrote:
2009/9/12 Chris Burns ch...@cloudtel.com:
Check out the variables ringback and transfer_ringback. The local
extension
Hello!
As I have a fax machine connected to an adapter that does T.38
(Grandstream HandyTone 502), I am playing with late codec negotiation
and proxy media. However, because late codec negotiation is a
profile-wide affair, I would like to know if there are any potential
drawbacks I should be
make samples
/b
On Sep 11, 2009, at 1:03 PM, Mark Sobkow wrote:
When I try to d a load mod_sofia, I get an error message indicating
that
Freeswitch can't find sofia.conf. There _is_ a sofia.conf.xml in the
autoload directory, which I _thought_ was the main sofia configuration
file. Do
sure, i have a full QA department who will take case of all possible cases.
Then it can be tested by our community.
Thank you.
On Fri, Sep 11, 2009 at 11:51 PM, Brian West br...@freeswitch.org wrote:
Also for tests make sure you fuzz test it also .. giving it invalid
data shouldn't crash ...
hi,
Was using this to listen in and most of the time it worked ok, but I had to
close and call in loads of times because sound went crap - but that's probably
skype - don't know.
Jan
From: gmar...@celliax.org
Date: Fri, 11 Sep 2009 18:12:34 +0200
To:
Kewl I have a fuzz test I do also thats automated that throws all
kinds of crazy stuff at all the api's.
/b
On Sep 11, 2009, at 2:52 PM, Muhammad Shahzad wrote:
sure, i have a full QA department who will take case of all possible
cases. Then it can be tested by our community.
Thank you.
great, can you share it with me?
Thank you.
On Sat, Sep 12, 2009 at 2:16 AM, Brian West br...@freeswitch.org wrote:
Kewl I have a fuzz test I do also thats automated that throws all
kinds of crazy stuff at all the api's.
/b
On Sep 11, 2009, at 2:52 PM, Muhammad Shahzad wrote:
sure, i
I'll dig it up this weekend and get you a copy of it.. its a perl
script that writes out some js that I run via jsrun
/b
On Sep 11, 2009, at 3:42 PM, Muhammad Shahzad wrote:
great, can you share it with me?
Thank you.
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Just thinking out loud. Wouldn't be
sofia_contact 180...@192.168.1.163 ?
jmesquita
On Fri, Sep 11, 2009 at 10:06 AM, Woody Dickson woodydick...@gmail.comwrote:
Hi,
I am having a strange problem here. sofia status shows that the user is
registered, but sofia_contact says the user is not
Hi there,
Here is my problem:
I'd like to set up a switchboard. I'm using python scripts called when a
call is coming. Here is my public.xml:
*extension name=toto
condition field=destination_number expression=[0-9]*
action application=python data=test.test2 /
/condition
This would require changes to the c code in mod_sofia. If you have a
patch to change this behavior (probably should address configuration
and authentication as well as this could be a denial of service path)
you can post it to http://jira.freeswitch.org.
Mike
On Sep 6, 2009, at 6:32 AM,
thanks for the info that it is a sip problem.
seems it should be doc in wiki to explain that how to configure freeswitch
so that client can connect from any interface,
cause not everyone play with freeswitch is a sip guru.
so thanks any way, i should learn more with sip and freeswitch.
On Sat, Sep 12, 2009 at 1:45 AM, Michael Jerris m...@jerris.com wrote:
Following up, did a bug get created for this issue?
Hello,
yes.
http://jira.freeswitch.org/browse/MODSOFIA-26
On Sep 5, 2009, at 2:40 AM, mayamatakeshi wrote:
On Fri, Sep 4, 2009 at 7:28 PM, mayamatakeshi
found it.
the correct typing in fs_cli is :/event plain CHANNELL_CREATE
freeswi...@internal /event plain CHANNEL_CREATE
+OK event listener enabled plain
2009/9/11 Anthony Minessale anthony.miness...@gmail.com
or from fs_cli
/events plain all
On Fri, Sep 11, 2009 at 8:21 AM, Brian
Hello,
I'm having trouble setting-up a T1 in Alaska to work with a Sangoma A101DE card.
I've followed the instructions on the Sangoma and FreeSWITCH websites, and a
support guy from Sangoma has dialed-in twice.
It could be an issue with the T1 itself, but I'm not sure how to rule that out.
I'm
On Fri, Sep 11, 2009 at 11:22 PM, Marc Orenberg m...@kasteris.com wrote:
I'm having trouble setting-up a T1 in Alaska to work with a Sangoma A101DE
card.
I've followed the instructions on the Sangoma and FreeSWITCH websites, and
a support guy from Sangoma has dialed-in twice.
Is this a PRI
I am anxious to provide my first real patch into FreeSWITCH and since this
looked like a good candidate, I looked at the code for a little while and I
have a few thoughts about the subject.
FreeSWITCH (mod_sofia) does not route chat messages to endpoints who are not
reachable (obviously). If you
2009/9/12 Chris Burns ch...@cloudtel.com:
There are a few ways you could go about dropping into a conference and playing
the song in from a separate channel.
Good idea :)
Thank.
On September 11, 2009 01:47:43 pm Dome Charoenyost wrote:
2009/9/12 Chris Burns ch...@cloudtel.com:
Check out
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