Hi,
I wonder, how can be implemented with opensips prepaid system with
service plans with included minutes. Can someone to give me some hints ?
Thanks,
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman
, DB, RADIUS, etc) and allows the billing to trigger the call
> termination from outside (like billing is keep computing costs and
> when there is no more credit, it notifies opensips to terminate the
> call) - again, you can use here the dialog module with the dlg_end_dlg
> command via
opensips.org/html/docs/modules/1.6.x/aaa_radius.html#id249105
>
> Regards,
> bogdan
>
> Dani Popa wrote:
>> Hi Bogdan,
>>
>> I know that opensips care just about SIP part. But my question is if
>> somebody tried to make this setup with call_control and opensips.
ol application for this. See
> http://callcontrol.ag-projects.com.
>
> Best Regards,
>
> Tijmen de Mes
> AG Projects
>
> Op 1/6/11 12:21 PM, Dani Popa schreef:
>> Hi,
>>
>> I wonder, how can be implemented with opensips prepaid system with
>> service
hi,
you have to use mediaproxy or rtpproxy for media timeout.
Dani
mancyb...@gmail.com wrote:
> Hi All,
>
> in a scenario where opensips routes calls from sip user agents to voip
> carriers:
> can you please confirm that the only way to be sure to prevent fraud false
> hangups
> is to force th
The auth is done at
> INVITE time, before sending the call to termination, while the ACC start is
> done at 200 OK INVITE, when the call is established.
>
> BTW, the acc module in opensips can automatically do RADIUS accounting (you
> do not need to do it manually).
>
>
>
rticular requirements .
>
> Adrian
>
> On Jan 6, 2011, at 5:40 PM, Dani Popa wrote:
>
>
>> Hi,
>>
>> I'm afraid i can not use call_control, because after DebitBalance or
>> MaxSessiontime, I can not categorize the call as belonging to a call
Hi,
I belive you want to use opensips without domain name. Check use_domain
value .
Dani
Chris Liu wrote:
Hi All,
I am new to opensips. I'd like to let users be able to login the system
by domain name and ip address.
For eg. our opensips server's ip is 123.123.123.123 , the d
Hi,
I get next error when i try to use aaa_radius
Jan 13 13:01:44 [10886] ERROR:aaa_radius:send_auth_func: radius
authentication message failed with ERROR
The opensips start normally with no error.
In script config i have:
modparam("aaa_radius", "radius_config",
"/etc/opensips_aaa_radius/radi
Hi,
For particular billing system you can use aaa_radius great opensips
module to integrate your billing system with wanted variables without
querying databases direct from opensips .
Dani
Andrew Philp wrote:
Hi All,
I am looking for a way to integrate calls going
throug
Hi,
I think you should make distinction between "call DND" and "presence
DND". DND for not receiving calls you can do from ACL or user
credentials. For presence, DND is indicated by UACs(also you cand use
fifo to change presence status for subscribers). Of course you can check
presence messages f
Hi,
If you use domain for auth , check if subscriber 1000 have domain
vmopensips1.skycomuk.com.
Dani
Gareth Blades wrote:
I have
just setup opensips and have a phone with a couple of lines which are
registered with opensips.
I have a problem when trying to place a call between the two
do you see in the system log any errors from the radius library ?
>
> Regards,
> Bogdan
>
> Dani Popa wrote:
>> Hi,
>> I get next error when i try to use aaa_radius
>>
>>
>> Jan 13 13:01:44 [10886] ERROR:aaa_radius:send_auth_func: radius
>> authe
Hi,
I'm trying to use aaa_radius for accounting, but aaa_radius modules
don't advertise Acct-Session-Type to radius. I miss something or
aaa_radius dont use acc engine from opensips ?
Thanks,
Dani
___
Users mailing list
Users@lists.opensips.org
http:/
Hi,
you can use usr_preferences avp_table and avp_load to load from db
incoming/outgoing barring rules.
in usr_preferences table you should have something like:
| 139 | | username | domain | 4122 |2 |
*barring_number* | 2011-01-11 12:19:31 |
define avp_aliases i:4122 named as y
nd_acct() ) or
> indirectly via the ACC module ? if you use it directly, you need to
> manually put all RADIUS AVPs in the requests, including the
> Acct-Session-Type (see the radius sets -
> http://www.opensips.org/html/docs/modules/1.6.x/aaa_radius.html#id249105)
>
> Regards,
>
What do you use for accountig ? radius_send_acct or acc ?
Dani
Roberto Santini wrote:
> Hi to all,
> I am new here and I have a problem with accounting on RADIUS.
> OpenSIPS sends start and stop accounting messages properly, but in case
> of update sends messages with Acct-Status-Type = 0, which
adius_send_acct.
>
> Roberto.
>
> On Tue, 2011-01-25 at 12:45 +0200, Dani Popa wrote:
>
>> What do you use for accountig ? radius_send_acct or acc ?
>>
>> Dani
>>
>> Roberto Santini wrote:
>>
>>> Hi to all,
>>> I am new here and I ha
Hi,
No, it is not necessary, but if you don't use rtpproxy, the device
behind NAT in some circumstances could not work.
Dani
Toyima Dias wrote:
> Hello,
>
> Is it necessary to have a rtpproxy installed into opensips server to
> be able to change/modify SDP headers in messages through the proxy
Hi,
Depending how you use db_mode for usrloc:
http://www.opensips.org/html/docs/modules/1.6.x/usrloc.html#id292952
If you user db_mode 0, then you have to restart the phone every time you
restart the opensips.
Dani
On 02/04/11 11:40, Toyima Dias wrote:
A dummy question...
Every time i r
Hi all,
How can i change timeout for radius_send_auth ? It is possible ?
Dani
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
You have a problem with opensips script. You set acct start flag many
times and radius try each time to insert in mysql the same start query,
but you can not because you have key defined on your table.
Dani
On 03/23/11 10:06, ha do wrote:
Hi list
i am test
opensips 1.6.4
freeradius from ag-p
you should enable sql in freeradius or use freeradius-xs(it have by
default sql enabled) provided by ag-projects
Dani
On 03/24/11 13:33, Tijmen de Mes wrote:
Hi,
You can enable sqltrace in freeradius to see what is wrong and try to
execute the queries manually.
Which version of freeradius ar
Hi,
How can i send MESSAGES only to users who are registered with exclusiv
User-Agent. I mean, A is registerd with User-Agent Jitsi and B is with
jitsi and Cisco phone. If A try to send messages to B, i need that
messages to be sent only to B on Jitsi, because cisco will replay with
"Not Impe
thanks,
Dani
On 04/04/11 18:55, Bogdan-Andrei Iancu wrote:
Hi Dani,
On 04/04/2011 06:46 PM, Dani Popa wrote:
Hi,
How can i send MESSAGES only to users who are registered with
exclusiv User-Agent. I mean, A is registerd with User-Agent Jitsi and
B is with jitsi and Cisco phone. If A try to
Hi,
Mediaproxy radius request does not populate Kbin and Kbout. Also i tried
to see sessions on port 25061 and also there callee_bytes and
caller_bytes are 0.
opensips:~# telnet localhost 25061
Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
sessions
[]
sessions
[{"from
e
machine. I wondering if is a network card driver issue or kernel
issue(if so, i'm dont know how to make troubleshooting, where should i
see the callee_bytes and caller_bytes in kernel stats).
Dani
On 04/18/11 10:43, Saúl Ibarra Corretgé wrote:
On 04/15/2011 02:42 PM, Dani
Hi,
I have a problem using b2b_init_request with "top hiding". When i
receive 200 ok for invite, opensips crash with
"ERROR:nat_traversal:__dialog_confirmed: FAKED reply - exit".
In core dump this is where opensips crash:
#0 get_source_uri (dlg=0xb2b4bc84, type=8, _params=0xb70b3c20) at
na
, Apr 20, 2011 at 8:11 AM, Dani Popa wrote:
Hi,
I have a problem using b2b_init_request with "top hiding". When i receive
200 ok for invite, opensips crash with
"ERROR:nat_traversal:__dialog_confirmed: FAKED reply - exit".
In core dump this is where opensips crash:
#0
at we also hit.. but still not the same.
Can you please paste the output of 'bt'**
<http://opensips.svn.sourceforge.net/viewvc/opensips/branches/1.6/modules/tm/uac.c?revision=7747&view=markup>
in gdb?
Regards,
--
Anca Vamanu
OpenSIPS Developer
On 04/20/2011 03:11 PM, D
I'm not able tu build mediaproxy on debian. Can someone give me a hint ?
Thanks,
Dani
root@test:/home/work/mediaproxy-2.4.4# ./setup.py build
running build
running build_py
creating build
creating build/lib.linux-i686-2.6
creating build/lib.linux-i686-2.6/mediaproxy
copying mediaproxy/__init__.
Hi,
yes, i was able to install it and run it, but i have some issues. I dont
have stream statistics: caller_bytes,callee_bytes,caller_packets and
callee_packets. Also, if i'm not sure if media timeout is working,
because i tried to simulate a hang call (in the middle of call, i
restart my har
883c080e30a8b9c ]---
Thanks,
Dani
On 04/21/11 13:51, Saúl Ibarra Corretgé wrote:
On 04/21/2011 12:44 PM, Dani Popa wrote:
Hi,
yes, i was able to install it and run it, but i have some issues. I dont
have stream statistics: caller_bytes,callee_bytes,caller_packets and
callee_packets. Also, if
On 04/21/11 14:13, Saúl Ibarra Corretgé wrote:
On 04/21/2011 01:06 PM, Dani Popa wrote:
sure,
Apr 21 06:06:41 test media-relay[4903]:
mediaproxy.mediacontrol.StreamListenerProtocol starting on 50012
Apr 21 06:06:41 test media-relay[4903]:
mediaproxy.mediacontrol.StreamListenerProtocol
OK,
Thanks,
Dani
On 04/21/11 15:14, Saúl Ibarra Corretgé wrote:
I'm not talking abut binding ports for streams, i'm talking about stream
packets and bytes info on telnet localhost 25060.
I meant the statisticas that get printed in syslog after the call is
closed.
[{"from_tag": "4fc7812
Hi,
Do you have any news with this issues ?
Thanks,
Dani
On Thu, Apr 21, 2011 at 3:31 PM, Dani Popa wrote:
> OK,
>
> Thanks,
> Dani
>
>
> On 04/21/11 15:14, Saúl Ibarra Corretgé wrote:
>
>>
>> I'm not talking abut binding ports for streams, i
Ok,
Thanks,
Dani
On Tue, May 3, 2011 at 10:00 AM, Saúl Ibarra Corretgé
wrote:
> On 05/02/2011 10:58 PM, Dani Popa wrote:
>
>> Hi,
>>
>> Do you have any news with this issues ?
>>
>>
> Unfortunately not. I didn't have time to go and fix this yet,
root@test:/opensips_1_6# opensips -V
version: opensips 1.6.4-2-tls (i386/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST,
SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_
Hi,
do you have news about this mediaproxy issues ?
Thanks,
Dani
On 05/03/11 11:52, Dani Popa wrote:
Ok,
Thanks,
Dani
On Tue, May 3, 2011 at 10:00 AM, Saúl Ibarra Corretgé
mailto:s...@ag-projects.com>> wrote:
On 05/02/2011 10:58 PM, Dani Popa wrote:
Hi,
Do yo
Hi Liviu,
What kernel do you have on running media-relay machine ?
Thanks,
Dani
On 05/26/11 11:14, Barsan Liviu wrote:
Hi,
With the python-gnutls update to 1.2.1 the mediaproxy works fine.
A suggestion: would be welcome a minimal install guide for
Ubuntu/Debian, for example I spent several d
when start openxcap, it try to take schema from www.w3.org/2001/xml.xsd
and www.w3.org doesn't responde.
I changed schemaLocation in
"/usr/local/pymodules/python2.6/xcap/appusage/xml-schemas/xcap-directory.xsd"
and pointed to local file.
Dani
On 06/06/11 03:01, duane.lar...@gmail.com wrote
greate
thanks,
Dani
On 06/06/11 16:18, Saúl Ibarra Corretgé wrote:
Hi Dani,
On Jun 6, 2011, at 3:07 PM, Dani Popa wrote:
when start openxcap, it try to take schema from www.w3.org/2001/xml.xsd and
www.w3.org doesn't responde.
I changed schemaLocation in
"/usr/local/pymodules
Hi all,
I looked on the internet for MOH with opensips as sip proxy(not b2b) and
other media servers (sems,asterisk,etc). The answers on internet was
that is not possible because SIP implementation and because
sems,asterisk are full implemented sip servers(invite from opensips to
media server
Hi,
I thought so, but I needed confirmation.
Thanks Adrian,
Dani
On 06/16/11 15:46, Adrian Georgescu wrote:
You cannot do this reliably unless you insert a B2BUA in the call flow.
Adrian
On Jun 16, 2011, at 2:11 PM, Dani Popa wrote:
Hi all,
I looked on the internet for MOH with
Hi all,
It is viable solution to use 30(1|2|5) redirect for REGISTER sip messages ?
Thanks,
Dani
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
As far as I know, opensips send 487, when receiving 200ok, when forking
On Dec 5, 2012 8:19 AM, "M.Khaled W Chehab" wrote:
>
> Dears ,
>
>
>
> How to send a 487 request terminated and drop the call directly if
the UA send a cancel ,since now I am sending 200 canceling to UA and send
a cancel
Hi,
I wondering if it posiible to add sdp on 180 ringing in order to play some
ringing tone. The ideea si that i want to play from rtpproxy with
rtpproxy_stream2uac/rtpproxy_stream2uas some music as ringback tone to
calling party if it's online.
--
Dani
I just want to play media on replay route in case of 18[013] reply, so i'm sure
the user was alerted if i got one of them, i'm pretty sure is not the case from
the link below and also inserted media is not a fake ringback.
Thanks anyway!
Dani Popa
On Feb 13, 2013, at 0:56, Daniel Go
you want to do is a 183 with early media,
> not just append an SDP to a 180.
>
> Good luck though:)
>
>
>
> -dg
>
>
> On Tue, Feb 12, 2013 at 3:28 PM, Dani Popa wrote:
>
>> I just want to play media on replay route in case of 18[013] reply, so
>> i'
log("MSILO: offline message NOT stored\n");
t_reply("503", "Service Unavailable");
};
}
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Five SIP clients with the same username.
Dani
On Fri, Feb 15, 2013 at 12:19 PM, Dani Popa wrote:
> Hi,
>
> Regarding msilo module and example from the documentation, one simple
> question:
>
> if i have 5 clients already registered and non of them know IM(message sip
d failure route is triggered when the transaction
> fails).
>
> So the final answer -> one time.
>
> Regards
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
>
> On 02/15/2013 12:22 PM, Dani Popa wrote:
>
> Five SIP clients
104.
> >
> >
> >
> >
> > From: Saúl Ibarra Corretgé
> > To: OpenSIPS users mailling list
> > Sent: Monday, February 18, 2013 3:10 PM
> > Subject: Re: [OpenSIPS-Users] msrp relay
> >
> >
> > On Feb 18, 2013, at 5:03 AM, nguyen khue wrote:
> >
> > > Hi all,
> > >
> > > How I can integrates msrprelay (msrprelay.org) with opensips to make
> File Transfer session between SIP end-points located behind NAT?. Please
> guide me.
> > > I tested file transfer between two SIP end-points in LAN and it worked
> successful.
> > >
> >
> > What problems did you ran into? Did you follow the installation guide
> http://msrprelay.org/projects/msrprelay/wiki/InstallationGuide ?
> >
> > Regards,
> >
> > --
> > Saúl Ibarra Corretgé
> > AG Projects
> >
> >
> >
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> --
> Saúl Ibarra Corretgé
> AG Projects
>
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
eb 18, 2013 at 4:50 PM, Saúl Ibarra Corretgé
wrote:
>
> On Feb 18, 2013, at 2:26 PM, Dani Popa wrote:
>
> > Hi,
> >
> > I think it's more helpful if you can give us calltrace in case of using
> msrp, sipproxy and of course 2 sip clients. Msrprelay it's a
mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
tp7585735.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _
__**_
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-**bin/mailman/listinfo/users<http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
ckets, LatePackets, LostPackets. I know,
some of you will recomand mediaproxy and it's not good for me, because i
chosed to use rtpproxy because, i can insert and record media in curent
stream. So the question is: there is any way to have such information at
the end of call?
Thanks,
--
(B)
(A)trying <-opensips <-trying(B)
(A)ringing <-opensips <-ringing(B)
(A)progress <-opensips
(A)200ok <-opensips <-200OK(B)
(A) ACK ->opensips ->ACK(B)
....
--
Dani Popa
___
(B)
(A)trying <-opensips <-trying(B)
(A)ringing <-opensips <-ringing(B)
(A)progress <-opensips
(A)200ok <-opensips <-200OK(B)
(A) ACK ->opensips ->ACK(B)
--
Dani Popa
___
Users mailing l
ckets, LatePackets, LostPackets. I know,
some of you will recomand mediaproxy and it's not good for me, because i
chosed to use rtpproxy because, i can insert and record media in curent
stream. So the question is: there is any way to have such information at
the end of call?
Thanks,
--
cc_db_request("200 Dialog Timeout", "acc");
}
}
Thanks,
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
any ideea ?
On Tue, Jun 18, 2013 at 7:10 PM, Dani Popa wrote:
> Hi all,
> I use acc with radius and when i set accountig flag in local_route i
> dont receive any accountig request on radius server. As I see local_route
> was hit twice on "dialog timeout" and i dont u
ry file in one or more places and opensips does not show it to you
> unless you have debug on.
>
> Regards,
> Qasim
>
>
> On Thu, Jun 20, 2013 at 11:25 PM, Dani Popa wrote:
>
>> any ideea ?
>>
>>
>> On Tue, Jun 18, 2013 at 7:10 PM, Dani Popa
).
> I suggest you to use the manual accounting in this case.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
>
> On 06/18/2013 07:10 PM, Dani Popa wrote:
>
> Hi all,
> I use acc with radius and when i s
>>> Willian Mazzardo
>>> Depto TI - SYSSVOIP
>>> www.syssvoip.com.br
>>> 55 3537 2030
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
azzardo
> Depto TI - SYSSVOIP
> www.syssvoip.com.br
> 55 3537 2030
>
>
> 2013/7/17 Dani Popa
>
>> set opensips peer to insecure=port,invite
>>
>>
>> On Wed, Jul 17, 2013 at 1:12 PM, Willian Mazzardo - SYSSVOIP <
>> will...@syssvoip.com.br&
xten => _X.,1,DeadAGI(a2billing.php)
>
>
> Willian Mazzardo
> Depto TI - SYSSVOIP
> www.syssvoip.com.br
> 55 3537 2030
>
>
> 2013/7/17 Dani Popa
>
>> what contex hit invite from opensips ?
>>
>>
>> On Wed, Jul 17, 2013 at 1:24 PM, Willian Mazz
There is any way to handle replay for sip keepalive OPTIONS packet when
using nathelper module ?
Thanks,
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
gards,
>
> Aamir Chougule
> Cell: 08097989101
> Skype-ID: aamir_ryu
>
> --- Sent from my BlackBerry ---
>
> -Original Message-
> From: Dani Popa
> Sender: users-boun...@lists.opensips.org
> Date: Fri, 13 Sep 2013 13:12:51
> To: OpenSIPS users mail
Hi all,
There is any way to check if Opensips instance have dialog in any state
defined by Replaces Header of new incoming call ?
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
____
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
r REGISTER.Probably you need to explicitly
test with the UACs you want use.
Regards,
Bogdan
On 06/20/2011 07:08 PM, Dani Popa wrote:
Hi all,
It is viable solution to use 30(1|2|5) redirect for REGISTER sip
messages ?
Thanks,
Dani
___
Users ma
Hi,
As far as i know it's hard to insert media from other sources in proxy
mode for situation like call hold or in call media insert. If you find a
solution, please let me know.
Dani
On 06/29/11 10:06, Barsan Liviu wrote:
Hi,
Yes, exactly. And obviously for this we want just one way stream
> ##}
> ##
> ##if (!db_check_to())
> ##{
> ##sl_send_reply("403","Forbidden auth ID");
> ##exit;
> ##}
>
> if (!save("location"))
> sl_reply_error();
>
>
ip:proxy@192.1.8.2.154")modparam("dispatcher", "ds_ping_interval",
> 30)modparam("dispatcher", "ds_probing_threshhold", 2)modparam("dispatcher",
> "ds_probing_mode", 1)modparam("dispatcher", "list_file",
> "/usr/local/etc/opensips/dispatcher.list")# modparam("dispatcher",
> "force_dst", 1) modparam("siptrace", "db_url",
> "mysql://root:Viamonte1621@localhost/opensips")modparam("siptrace",
> "enable_ack_trace", 1)modparam("siptrace", "trace_on",
> 1)modparam("siptrace", "table", "sip_trace")modparam("siptrace",
> "trace_flag", 22) #modparam("acc", "log_level", 1)#modparam("acc",
> "log_flag", 1)#modparam("acc", "db_url",
> "mysql://root:Viamonte1621@localhost/opensips") route{
> setflag(22);setbflag(22);sip_trace();if (
> !mf_process_maxfwd_header("10") ){
> sl_send_reply("483","To Many Hops"); drop();};
> if (is_method("OPTIONS")) {options_reply();
> exit;}ds_select_dst("1", "0");if
> ($retcodexlog("[Redmond] Service full\n");
> sl_send_reply("500","Service full");exit;}
> forward();#t_relay();#sip_trace();} failure_route[1]
> {if (t_check_status("(408)|(5[0-9][0-9])")) {
> ds_mark_dst();if (ds_select_dst("1", "0"))
> {forward();} else
> { t_reply("503", "Service
> Unavailable");}}} onreply_route {
> setflag(22);setflag(22);sip_trace();} ThanksDiego
> >
> >
> > >
> >
> >
> > >
> >
> >
> > > > >
> >
> >
> > >
> >
> >
> > >
> >
> >
> > > > >
> >
> >
> > >
> >
> >
> > >
> >
> >
> > > >
> >
> >
> > >
> >
> >
> > >
> >
> >
> > > >
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
--
Dani Popa
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
HI,
first
aaa_radius_auth and specific sql procedure in sql server.
the second
asterisk/freeswitch load balncing
Dani
On 07/12/11 17:06, duane.lar...@gmail.com wrote:
For your first question would this work?
http://www.ag-projects.com/projects-products-96/535-call-control
For your second q
Hi,
How can i solve this kind of problems ? Opensips doesn't crash, but it
not respond to any sip requests.
Aug 3 07:36:48 test /usr/local/sbin/opensips[29094]:
WARNING:core:fm_malloc: Not enough free memory, will atempt defragmenation
Aug 3 07:36:48 test /usr/local/sbin/opensips[29094]:
Hi all,
How can i remove all sip video body headers regardin video. Should i
remove any line from body after "m=video", or how. Please give me a
hint, if you have.
Thanks,
Dani
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.o
wrote:
Hi Dani,
Why would you do that? If you don't want to allow video, you can
simply replace the video port in the "m=" line with 0.
Regards,
Razvan Crainea
OpenSIPS Developer
On 04.08.2011 16:58, Dani Popa wrote:
Hi all,
How can i remove all sip video body headers regard
, Razvan Crainea wrote:
Hi Dani,
It seems you are out of memory. What version of OpenSIPS are you using?
Regards,
Razvan Crainea
OpenSIPS Developer
On 04.08.2011 16:07, Dani Popa wrote:
Hi,
How can i solve this kind of problems ? Opensips doesn't crash, but
it not respond to any sip req
Ok,
thanks for quick response.
Dani
On 08/04/11 18:26, Vlad Paiu wrote:
Hello,
Is it possible that you upgrade to 1.7 ? It is possible that this
issue was fixed in the latest OpenSIPS version.
If not, go to Makefile.defs, uncomment the line with
-DDBG_QM_MALLOC \
and comment the line w
0.
[1] http://www.opensips.org/html/docs/modules/devel/textops.html#id293910
Regards,
Razvan Crainea
OpenSIPS Developer
On 04.08.2011 18:03, Dani Popa wrote:
Hi,
In fact, i have some problems with one of my pstn gw's that send "400
Incorrect content length", i think, because
Hi,
it is somehow that username from sip uri to be non case sensitive when
we talk about presence and xcap storage? I mean, if userA add userB, in
his contact list, i need userA to be able to add userB even he add
him(type) as USERB.
Dani
___
Us
Hi,
Ok, but also, registrar module support non "case sensitive" sip username.
--
Dani Popa
On 8/5/11 11:40 AM, Vlad Paiu wrote:
Hello,
What you're asking for is against the RFC 3261 URI comparison rules,
which states that comparison of the userinfo part of the URI shoul
Hi,
When using m_store("$ru") the SIP messages sent back to sender have
default server_header and not the one i rewrite it.
Dani
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
RFC compliant.
Regards,
Bogdan
On 08/05/2011 07:30 PM, Dani Popa wrote:
Hi,
Ok, but also, registrar module support non "case sensitive" sip
username.
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-b
Hi,
Where should i find memory dump ? I have something in logs about memory.
I'll attach an file. Please let me know if this is what you need.
I also increased PKG_MEM_POOL_SIZE = 8 *1024 * 1024, and shared mem to
256, and also updated opensips 1.6.4 to latest svn revision, i think.
root@
Hi again,
i also saw that i compiled opensips with libxmlrpc-c3-dev and
libxmlrpc-c3 and i was warned somewhere that i'll compile it on my own
risk. Now i removed libxmlrpc-c3-dev and libxmlrpc-c3 and i compiled
with libxmlrpc-c++4-dev without warnings.
Let's see what we will get!
Thanks,
Hi,
True, i changed wrong the Makefiles.defs.
I dont know if you need this:
if i change Makefile.defs as:
DEFS+= $(extra_defs) \
.
.
.
.
-DCHANGEABLE_DEBUG_LEVEL \
#-DF_MALLOC \
-DDBG_QM_MALLOC \
#-DDBG_F_MALLOC \
opensips will not be compiled with -DDBG_QM
get only the memory dump without all runtime logs from mem
debugger.
Regards,
Bogdan
On 08/19/2011 01:13 PM, Dani Popa wrote:
Hi,
True, i changed wrong the Makefiles.defs.
I dont know if you need this:
if i change Makefile.defs as:
DEFS+= $(extra_defs
to get only the memory dump without all runtime logs from mem
debugger.
Regards,
Bogdan
On 08/19/2011 01:13 PM, Dani Popa wrote:
Hi,
True, i changed wrong the Makefiles.defs.
I dont know if you need this:
if i change Makefile.defs as:
DEFS+= $(extra_defs
the memory dump without all runtime logs from mem
debugger.
Regards,
Bogdan
On 08/19/2011 01:13 PM, Dani Popa wrote:
Hi,
True, i changed wrong the Makefiles.defs.
I dont know if you need this:
if i change Makefile.defs as:
DEFS+= $(extra_defs) \
.
.
.
.
-DCHANGEABLE_DEBUG_LEVEL
in the DBG support, set:
memlog=6
memdump=1
in order to get only the memory dump without all runtime logs from mem
debugger.
Regards,
Bogdan
On 08/19/2011 01:13 PM, Dani Popa wrote:
Hi,
True, i changed wrong the Makefiles.defs.
I dont know if you need this:
if i change Makefile.defs
Hi,
I think you could use dialog profile, but not sure.
Dani
On 08/19/11 23:17, Robert Thomas wrote:
Hi,
I have a load balancer module to distribute calls among my
Gateways. I can use the lb_list command to see the active calls per gw, but I
would like something similar to graph my customer
Hi,
should "412 Conditional Request Failed" for PUBLISH (if SIP-If-Match
doesn't match) work with opensips 1.6.4 ?
Thanks,
Dani
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
compiled on 05:48:37 Aug 19 2011 with gcc 4.5.2
Thanks,
Dani Popa
On 08/19/11 17:31, Bogdan-Andrei Iancu wrote:
Hi Dani,
You can not have comments in multi-line assignments
So, instead of
DEFS+= $(extra_defs) \
.
-DCHANGEABLE_DEBUG_LEVEL \
#-DF_MALLOC
Thanks,
Dani
On 08/19/11 18:01, Bogdan-Andrei Iancu wrote:
Hi Dani,
In your case opensips will act as UAC (not server), so you need to
define your custom user_agent_header:
http://www.opensips.org/Resources/DocsCoreFcn17#toc96
Regards,
Bogdan
On 08/16/2011 03:12 PM, Dani Popa wrote
, set:
memlog=6
memdump=1
in order to get only the memory dump without all runtime logs from mem
debugger.
Regards,
Bogdan
On 08/19/2011 01:13 PM, Dani Popa wrote:
Hi,
True, i changed wrong the Makefiles.defs.
I dont know if you need this:
if i change Makefile.defs as:
DEFS
1 - 100 of 130 matches
Mail list logo
