We have tried setting both effective_caller_id_number and
origination_caller_id_number:
session1.originate(session1,{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317,15);
but the problem still exists. The solution we have found for the case
when we originate two calls,
Hello list,
I'm trying to track down a seg fault issue with a fs Revision: 11489
Here is the backtrace pastebin:
http://pastebin.freeswitch.org/7009
but before digging the dump I would like to understand: am I the only
one having error like this in fs console:
Error in my_thread_global_end(): 16
I have seen that error myself, however I assumed it was due te me
hanging up other cals from the api_hangup_hook of a related call.
I use this to set a master call in a conference so that if it hangs up
all calls in the conference hangup.
On Wed, Feb 04, 2009 at 12:03:12PM +0100, Cavalera
Hello
I'm trying to make outbound calls through my gateway provider.
My calls got rejected and I asked them why.
Apparently I need to use 5060 as source port, since they validate both my IP
and the port that the messages come from.
Is this possible with freeswitch? If so, what config settings
If you still get a crash on SVN trunk please post the bt even if you think
it's the same, since it won't be
exactly the same, the line numbers etc will be accurate with our development
code making it easier to debug.
On Wed, Feb 4, 2009 at 12:38 AM, shehzad p pmh...@gmail.com wrote:
Hi
Where did you learn how to use js this way?
session.originate is being misused here and is depricated and may be
removed.
the first arg to session.originate is either undefined or a *different*
session (the a leg)
session1 = new Session();
session1.originate(undefined,
Hello,
I'm trying to compile a brand new fs on a clean system.
Revision: 11630
After the usual ./bootstrap.sh
./configure --enable-core-odbc-support
I was getting this at make
http://pastebin.freeswitch.org/7011
so
i cd into utils under libs/sofia-sip/utils
issued a doxygen -u
but still getting
Thanks Anthony,
the js snippets are very instructive.
A couple of points:
1. The code with apiExecute does not work (local phone is connected, but
after picking up it hungs up immediately), other examples are working
fine.
2. It does not show how initiate external call without existing session.
I'm behind NAT. Is it FS that picks the random port, or the FW?
I've mapped port 5060 to the freeswitch ip in my FW.
On Wed, Feb 4, 2009 at 3:14 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
if you are not behind any nat then as long as you run your profile on 5060,
the source
On Wed, Feb 4, 2009 at 8:20 PM, Jonas Gauffin jonas.gauf...@gmail.comwrote:
I'm behind NAT. Is it FS that picks the random port, or the FW?
the FW
I've mapped port 5060 to the freeswitch ip in my FW.
thats inbound, now you need to tell your firewall to nail port 5060 on the
outbound side
It is the firewall. Most consumer firewalls allow mapping inbound ports
(probably what you describe). I don't know of any that do outbound
mapping. Linux or *bsd firewalls should be able to do what you want.
I'm sure a cisco with IOS could but it has been ages since I've played
with that.
On
Its the firewall... some fw's will give you port 5060 if there's nothing
else in the network using 5060, but others will randomize it always.
-Ray
Jonas Gauffin wrote:
I'm behind NAT.
Is it FS that picks the random port, or the FW?
I've mapped port 5060 to the freeswitch ip in my FW.
On
freeswitch-users-boun...@lists.freeswitch.org wrote:
I have seen that error myself, however I assumed it was due te me
hanging up other cals from the api_hangup_hook of a related call.
I use this to set a master call in a conference so that if
it hangs up
all calls in the conference hangup.
can you make current to rule out any issues with outdated code and maybe
describe what you are using in FS such as scripting langs or anything else
that was not enabled by default.
2009/2/4 Cavalera Claudio Luigi claudio.caval...@italtel.it
freeswitch-users-boun...@lists.freeswitch.org wrote:
can you press f8 for debug and try that apiExecute and post the results?
On Wed, Feb 4, 2009 at 8:46 AM, Jacek Sokulski jsokul...@dotsystems.plwrote:
Thanks Anthony,
the js snippets are very instructive.
A couple of points:
1. The code with apiExecute does not work (local phone is connected,
This should now be fixed in trunk in revision 11632. Can you please test
and confirm.
Mike
On 2/4/09 9:27 AM, Cavalera Claudio Luigi claudio.caval...@italtel.it
wrote:
Hello,
I'm trying to compile a brand new fs on a clean system.
Revision: 11630
After the usual ./bootstrap.sh
Hi,
I have a problem in SIP registration (authentication) with FreeSWITCH server.
The SIP messages are:
recv 292 bytes from udp/[209.82.10.250]:3458 at
16:35:24.758862:
REGISTER
sip:209.82.10.235 SIP/2.0
That's the reason, the missing params.
The client has a bug in it.
On Wed, Feb 4, 2009 at 10:17 AM, Ali Al-Rubaie kerrada2...@yahoo.comwrote:
Hi,
I have a problem in SIP registration (authentication) with FreeSWITCH
server. The SIP messages are:
recv 292 bytes from
Anthony Minessale napsal(a):
What does it look like if you serve the directory from the static xml
file out of curiosity.
Well, I write all user infos into static xml files loaded at startup :-)
For the first try (without tuning, see below) I can't go beyond 220
reg/s - it is just about
I'll make sure the substance of this is in the wiki and I'll look for
references to the deprecated way and remove those.
-MC
On Wed, Feb 4, 2009 at 6:09 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
Where did you learn how to use js this way?
session.originate is being misused here
What I did was the following:
First, I sent the playback command:
call-command: execute
execute-app-name: playback
execute-app-arg: filename
Then I send uuid_record (Sorry, it was not Record command):
api uuid_record uuid start filename 120
I also tried replacing the playback command
mod_limit is just another lego brick in the set... You can put it in
the dialplan before you go in our out a gateway and limit based on the
realm... its no different for registered users either... I get the
feeling you're trying to make this more complicated then it really is.
/b
PS when
On Wed, Feb 4, 2009 at 9:56 AM, Gopalakrishnan A.N sai...@gmail.com wrote:
Hi,
Its a awesome. Can the packet capturing be done with event socket?
Not at this time. Would require some additional programming. Are you
up for the task? ;)
-MC
--
Thank you with regards,
Gopal,
Hi Anthony,
I have been seeing this message for the for a couple of weeks now. And
as you know(you asked me to) I have been keeping up to date.
So it is definately not only a recent change. I use JS with odbc and
core support. And also xml_rpc. I will do some more testing tommorow to
see if I
Hi Guys,
Excuse my ignorance, but I'm just starting with FS.
I've loaded FS onto one of our servers in a datacenter. I'm registering
with our PSTN breakout provider just fine, but I'm a little confused
about internal/external.
Given that we have no internal clients, as they're all
Hi Guys,
Need a little help here; I connect to my PSTN provider via the LAN,
Question: As the provider authenticates on IP, how do I not send a
password? In the .xml file if I remove the password entry it complains
Secondly, the contact should be my local address, not the public one.
Don't let the names of the profiles confuse you... they are just
names. internal is on port 5060 has auth on... external is on 5080
and doesn't' have auth on and lets all calls into the public context
without auth.
Also when you post to the mailing list do not hijack a thread.
Hijacking
Do apologise about the hijacking,
Question: My ISP sends inbound calls via 5060, so it seems I need to
renumber the ports, but that leaves my SIP end points who authenticate
also needing 5060, can they be combined?
Regards,
-Original Message-
From:
Hi Guys,
Need a little help here; I connect to my PSTN provider via the LAN,
Question: As the provider authenticates on IP, how do I not send a
password? In the .xml file if I remove the password entry it complains
Secondly, the contact should be my local address, not the public one.
If your ITSP requires you to come from 5060 on your request then they
are seriously broken. But yes you can move the ports around on the
profiles or turn auth to false on the internal profile if you don't
require any digest auth or phones registering.
/b
On Feb 4, 2009, at 4:50 PM, Nik
Well try as I might, I can't connect to that server, others are fine,
but I get DNS pool errors
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Brian West
Sent: 04 February 2009 23:00
You can reverse the ports or try to get your provider to send to 5060 or you
can bind an additional IP and have the external profile listen there
K
From: Nik Middleton nik.middle...@noblesolutions.co.uk
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Wed, 4 Feb 2009 23:07:58 -
To:
Hello,
I'm having it on both Fedora and Ubuntu boxes:
2009-02-05 01:26:27 [ERR] mod_xml_cdr.c:290 mod_xml_cdr_load() Open of
xml_cdr.conf failed
2009-02-05 01:26:27 [CRIT] switch_loadable_module.c:839
switch_loadable_module_load_file() Error Loading module
/usr/local/freeswitch/mod/mod_xml_cdr.so
Make sure your config file is installed and issue a reloadxml then
load mod_xml_cdr
/b
On Feb 5, 2009, at 1:33 AM, Shelby Ramsey wrote:
Hello,
I'm having it on both Fedora and Ubuntu boxes:
2009-02-05 01:26:27 [ERR] mod_xml_cdr.c:290 mod_xml_cdr_load() Open
of xml_cdr.conf failed
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