of calls. Just to be clear, I do not have
> anything specified in my cfg file.
>
> Thanks!
> Alex
>
> Hi Alex,
The short answer is no :)
Why you think that you hit any limits? Did you see something in the logs?
- Laszlo
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On Tue, Mar 27, 2018 at 3:31 PM, Rodrigo Pimenta Carvalho wrote:
> Hi.
>
>
> When a peer invites another one to a call, there are calling and ringing
> tones for these peers.
>
> My SIP agents let these tones execute during 2 minutes. After this, the
> call is terminated, if
On Wed, Aug 9, 2017 at 3:35 PM, wrote:
> Hi
>
>
>
> PHP 5.4.16 (cli) (built: Aug 5 2016 07:50:38)
>
> Copyright (c) 1997-2013 The PHP Group
>
> Zend Engine v2.4.0, Copyright (c) 1998-2013 Zend Technologies
>
You might need to install
am.
>
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Hi Eric,
The default opensips config is suitable for testing your scenario.
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http
/ratelimit.html#id293757
(the first example)
Also make sure that you set the timer_interval
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config.
Thank you very much!
Regards,
Markus
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need to open other ports/protocols, too?
With kind regards,
Markus
Am 27.01.2015 um 19:21 schrieb Laszlo:
On Sun, Jan 25, 2015 at 10:00 PM, Markus Klaus Schäffauer
mar...@mkschaeffauer.de wrote:
Hello,
now I have OpenSIPS running and two VoIP-Phones registered via my
OpenSIPS
/
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Try running it under PHP 5.3.x, or you can remove ampersands from 3-4 lines
if you know PHP well :)
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Which version of python-application you using?
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the function expects 4 mandatory parameters, not 3
- Laszli
2014.03.18. 18:04, Alcindo Schleder alcindoschle...@gmail.com ezt írta:
I'm trying to use the function and check_address get an error compiling
the script.
Script line 390.
if (check_address(1, $si, $sp)) {
...@lists.opensips.org [mailto:
users-boun...@lists.opensips.org] *Em nome de *Laszlo
*Enviada em:* terça-feira, 18 de março de 2014 14:24
*Para:* OpenSIPS users mailling list
*Assunto:* Re: [OpenSIPS-Users] Error in Module Permissions
the function expects 4 mandatory parameters, not 3
If you added an usrloc entry, then you will need to use some kind of
locaton lookup and then route accordingly.
- Laszlo
2014-03-10 11:03 GMT+01:00 Mike Claudi Pedersen mike.peder...@ipnordic.dk:
im trying to establish connection between 2 phones
user1: 43384001
user2: 43384002
i have
looks broken for me, at least for the lr54.204.19.238
part... as it is not following the ;name=value convention
Maybe you doin something weird in your config :)
-Laszlo
2013/12/20 Zoho Junk m...@mmilburn.com
Greetings,
I have OpenSips installed on an Amazon EC2 with an elastic IP
We using it in a combination with the dialplan module.
But it's quite complicated, as LRN dipping, inter and intrastate routing is
used here (together with some profit protection).
- Laszlo
2013/12/9 Nick Cameo sym...@gmail.com
Hello Everyone,
I am currently working on an LCR for A-Z using
Maybe you looking for this one?
http://www.opensips.org/html/docs/modules/1.10.x/options.html
-Laszlo
2013/11/27 Nick Cameo sym...@gmail.com
Hello Everyone,
Is there a correct way to handle OPTIONS requests using OpenSIPS?
Our server returns 484s to OPTION requests, and causes the carrier
and ncurses-devel :)
- Laszlo
2013.11.08. 13:52, Schneur Rosenberg rosenberg11...@gmail.com ezt írta:
Install ncurses
On Nov 8, 2013 12:16 PM, Vishnu Vardhan vishnuv...@gmail.com wrote:
Hi,
When i am trying to configure Opensips in Centos 6.4 i am getting this
error messgae
I can't comment on astpp, never used it on a production system.
BTW, if you have unique needs, did you try to contact AG Projects
directly? Adrian and others are subscibers in this mailing list, so they
can help directly to you with commercial support, since they maintaining
cdrtool.
- Laszlo
The answer is not that simple by just saying yes or no.
We started with cdrtool in the beginning, but now only the retail customers
are handled by cdrtool.
The cdrs are already in radacct, you can use custom queries to retrieve the
info you need, and then extend cdrtool to your taste.
- Laszlo
What if you simply drop the 180 in the onreply_route?
-Laszlo
2013/9/23 Jeff Pyle jp...@fidelityvoice.com
Hello,
I have one particular PSTN call flow that causes a 183 with SDP, then a
180 without SDP prior to 200 OK. Some of my customer endpoints don't
handle the 180 properly after
Can you xlog $avp(callfwd) and $ru?
-Laszlo
2013/9/6 Tấn Phát Ngô phat...@gmail.com
Hi all,
I want to forward call to client C when client A calls client B in a
specified time, example: after 17h30specified time is optional.
I implement as Building Telephony Systems with OpenSIPS
You can try adding /etc/init.d/opensips start to /etc/rc.local
- Laszlo
2013/8/29 Danny Dias ing.diasda...@gmail.com
I think that the problem is that OpenSIPS is starting before MySQL, that's
why opensips stops:
*Aug 28 23:15:26 opencomm01 opensips: INFO:core:daemonize: pre-daemon
process
Hi Jeff,
codec_delete_except_re(PCMU|tel);
should do the trick.
-Laszlo
2013/8/16 Jeff Pyle jp...@fidelityvoice.com
Hello,
This is on OpenSIPS 1.9.
I want to remove all audio codecs except PCMU from an SDP. I have the
following config:
if (codec_exists(PCMU
Hi,
Did you install the db_mysql module in opensips?
-Laszlo
2013/8/12 troxlinux xserverli...@gmail.com
Hi , I tried to install opensips on centos 6.4 X64 , and I have all
sources installed mysql
opensipsdbctl create
ERROR: could not load the script in /usr/local/lib64/opensips
Hmm, and db_mysql.so can be found under /usr/local/lib64/opensips/modules/ ?
-Laszlo
2013/8/12 troxlinux xserverli...@gmail.com
Yes , look Makefile
exclude_modules?= b2b_logic cpl-c xmpp rls mi_xmlrpc xcap_client \
db_oracle db_berkeley \
osp perl db_perlvdb
Then the module is not installed ;)
Try to recompile opensips with the following commands.
make include_modules=db_mysql modules
make include_modules=db_mysql install
And check if db_mysql.so is installed...or not.
-Laszlo
2013/8/12 troxlinux xserverli...@gmail.com
not really ..
[root
create.
-Laszlo
2013/8/12 troxlinux xserverli...@gmail.com
Thnk Laslo , but shows me another error and I have set the mysql user and
password in opensipsctlrc
opensipsdbctl create
MySQL password for root:
INFO: test server charset
Warning: Using a password on the command line interface
Maybe I have missed something, but did you try without TLS? So try it with
TCP first. If that goes well, then you can start testing with TLS.
-Laszlo
2013/8/7 debelizmaj demj...@gmail.com
I tried with csipsimple client, tls+trpproxy+opensips.
But no rtp stream.
Don't have idea what to do
entry.
-Laszlo
2013/8/5 Jeff Pyle jp...@fidelityvoice.com
Hello,
I'm using rtpproxy 1.2.1 with Opensips 1.9.0 r9980. The configuration is
multihomed, one public interface and one private. Calls that come from the
public side t_relay() to the private one and vice versa with rtpproxy
http://openser.org/pipermail/users/2013-May/025705.html
maybe this can give you an idea..
-Laszlo
2013/8/3 mayamatakeshi mayamatake...@gmail.com
Hello, can anyone give me an example of how to call drouting function
is_from_gw ?
http://www.opensips.org/html/docs/modules/1.9.x
Bogdan,
Will you going to record the event and make it available for those guys,
who don't have time to attend?
2013.08.01. 20:03, Bogdan-Andrei Iancu bog...@opensips.org ezt írta:
**
*OpenSIPS Summit, 5th of August 2013, Chicago, US, collocated with
ClueCon Conference
*
*Norman Brandinger
It is not well documented, but it's known to be working with usrloc,
dialog, permissions and drouting.
You may need to try it out with other modules, it won't be hard to verify :)
-Laszlo
2013/7/23 Duane Larson duane.lar...@gmail.com
I know that with the NoSQL modules you have the Exported
in the patch file.
-Laszlo
2013/7/3 ㄨ冷se灬 291490...@qq.com
dear all:
I update my freeradius from v1.1.8 to v2.2.0.
when I use v2.2.0 ,I can’t see unsuccessfully calls(like
cancel,etc)
and radius got this log:
rad_recv: Accounting-Request packet from host
Hi Jens,
Can you make a ngrep trace?
That way we can see more things about what is going on...
Do you see anything wrong in your syslog during the call?
-Laszlo
2013.06.22. 19:39, Jens Sauer sauer.j...@yahoo.de ezt írta:
Hello,
since a month i try to solve my rtp problem. I´m quite new
Do you use create_dialog(B); ?
-Laszlo
2013/6/24 qasimak...@gmail.com qasimak...@gmail.com
Hi,
I am using radius accounting and during that accounting i calculate the
maximum amount to call duration. I am setting dialog timout_avp in my route
but the call doesn't hangup. From
useragent.
And then you can run your notify stuff or you can do anything with the call.
Just an idea.
-Laszlo
2013.06.22. 3:34, microx acmic...@gmail.com ezt írta:
Hi all,
I encounter a particular case as follows. I set up a SIP proxy server using
OpenSIPS. An iOS VoIP app is developed to make
BTW, looks like you doing uac_replace_to twice. once in global, and once in
a per-branch route.
You can do it only once, I'm sure you see something about this in the
syslog.
try what Flavio suggested, so do these changes only in the branch route.
-Laszlo
2013/6/19 M.Khaled W Chehab kche
:
http://lists.opensips.org/pipermail/users/2010-June/013200.html
Took 40 seconds to find.
-Laszlo
2013/6/19 M.Khaled W Chehab kche...@icucall.com
Please can you show me what do you mean by coding it ,what do you mean
branch route for example is it route (6) I am using , you can find
with end_media_session() upon receiving
the BYE.
-Laszlo
2013/6/19 M.Khaled W Chehab kche...@icucall.com
Hi Laszlo
** **
I can see this error in the syslog
error: Got `remove' command from OpenSIPS for unknown session with call-id
`ca8ccd95-d81511e2-bebfd582-e4ab...@xx.xx.xx.xx
Cool!
If you checked the link what I have sent, then you can easily see the logic
how it can be implemented.
-Laszlo
2013/6/19 M.Khaled W Chehab kche...@icucall.com
Hi Laszlo
** **
I am searching since yesterday,as I inform you that I use uac_restore_to()
and I set iuac_replace_to
what happens if you manually set $avp(can_uri) to a fixed value? like to
, and define this destination in cdrtool?
2013.06.11. 18:08, Leonardo Uzcudun uzcud...@yahoo.it ezt írta:
Did it but still not working. I'm still receiving Maxsessiontime=unlimited
and DetsId=none :(
Is there any
Hmm, this sounds challenging :)
I'm working on a custom opensips user portal and admin interface at the
moment, and it is a nice to have feature here too.
cdrtool is also used here, so I'll play with this feature on my dev server.
Let's see if there can be a solution, or not.
-Laszlo
no, it's not running at all. Can you look at the relevant log entries (if
any) in the syslog during you start running rtpproxy? Also, what is the
command you used for starting rtpproxy?
-Laszlo
2013/6/3 Nandini madhu sermj2...@gmail.com
Thanks for your reply.
i have got the following output
http://qmd-training.com/tzn/kdtludprbbtcxhacgcsrga.dfbsxx
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svnrevision: 2:9980
@(#) $Id: main.c 9790 2013-02-15 10:14:34Z bogdan_iancu $
main.c compiled on 00:52:55 Apr 30 2013 with gcc 4.4.6
-Laszlo
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looks like it's unable to parse the password for 'localhost' in the servers
file.
(radiusclient-ng/servers is the file).
what host you have in that file? localhost or 127.0.0.1?
if 127.0.0.1, change it to localhost and try again.
-Laszlo
2013/4/11 leo uzcud...@yahoo.it
Hello:
If i
+1 on this
2013.03.23. 19:02, dotnetdub dotnet...@gmail.com ezt írta:
Hi AG,
The new version looks amazing. Is there any reason to leave out the
'view sip trace as plain text' option?
The SIP trace formatted is lovely and all but the plain text option is
much better so you can scan trace
INT(10) UNSIGNED NOT NULL
was added.
plus there is a new index:
CREATE INDEX hash_idx ON dialog (hash_entry, hash_id);
And in the version table, the table_version is now 8.
-Laszlo
2013/3/18 qasimak...@gmail.com qasimak...@gmail.com:
Also since you copied script from 1.8 you should consider
('dialog', 'dlg_match_mode', 1)
modparam('dialog', 'db_mode', 2)
Regards,
Peter
On 18 Mar 2013, at 11:46, Laszlo las...@voipfreak.net wrote:
Also check the table schema of the dialog module, see
dialog-create.sql under /scripts/mysql
There are differences between 1.8 and 1.9.
dlg_id BIGINT(10
you need to install the ncurses-devel package via yum
-Laszlo
2012/11/22 Carlos Cruz car...@nbtbizcapital.com:
I really don't know why it needs to be so difficult to install a piece of
software?
I'm trying to install OpenSIPS on CentOS 5.8 below are the errors I get:
I've done quite
It won't do anything if you used engage_media_proxy(), see
http://www.opensips.org/html/docs/modules/1.6.x/mediaproxy.html#id250230
2012/10/11 Binan AL Halabi binanalhal...@yahoo.com:
Hi,
# BYE processing
if (method==BYE) {
end_media_session();
}
// Binan
It could be a good idea to list this info at opensips.org site, as this
problem is popping up many times.
-Laszlo
2012.06.06. 21:16, Brett Nemeroff br...@nemeroff.com ezt írta:
It's possible that your default route in syslog also is picking up local0.
For good measure I'd also stick a - before
maybe you also facing this problem:
http://opensips.org/pipermail/users/2011-November/019775.html
-Laszlo
2012/6/5 yassine.ab yassine...@voila.fr:
Hi all,
when I run:
http://localhost/CDRTool/network_status.phtml
I have this error:
Error connecting to tcp://127.0.0.1:25061: (Could
2011/3/11 Toyima Dias toyim...@gmail.com
so, the RPID will identify the originator of the call (in terms of the
PSNT), right? what if i do not set any RPID or PAI, what will be presented
as teh originator of the call on the pstn destiny?
Then it will be taken from the From
($fU)
Try to increase the verbosity in the log and see what happens.
You should be able to push more cps than 5, unless you have something wrong
in you cfg, or your vendor limited your gw, etc.
-Laszlo
2011/2/21 chris ch...@ghosttelecom.com
Hi,
Can anyone help with a performance problem.
I
Hi Adrian,
2011/2/10 Adrian Vasile y...@opennet.ro
Hello all,
Maybe it has happened to you too.. I've got a couple of cisco spa504g
everything is fine with them, registering, calling out, but there seems to
be a problem with the calling in feature..
When I try to call the spa's all they
can be done with linux HA easily.
also try to read this:
http://www.smartvox.co.uk/astfaq_ha_failover_ideas.htm
2011/1/26 Toyima Dias toyim...@gmail.com
HEllo my friends,
I wonder if someone has done this before? i would like to implement 2
OpenSIPS servers with redundancy, something like
also try to increase this in config.h, then recompile:
#define PKG_MEM_POOL_SIZE 1024*1024
set it to 2*1024*1024 for testing
2011/1/24 Denis Putyato denis7...@mail.ru
Hello!
I have such error messages in syslog while using 1.6.4
Jan 24 09:40:04 opensips
Hmm, it's like Ferrari owners talking about which one is better: Volkswagen
or Toyota :)
2010/12/10 Aloysius Lloyd lloyd.aloys...@gmail.com
Paul,
I do not quite understand what is find me doing with NAT
Thanks
Lloyd
On Fri, Dec 10, 2010 at 10:11 AM, Jeff Pyle
(
http://www.opensips.org/html/docs/modules/1.6.x/benchmark.html)
Nobody can answer your question other than you :)
-Laszlo
2010/12/8 Vic Jolin victor.jo...@gmail.com
Can you give me a glimpse of your configuration? maybe just the basic
routing you have?
On Fri, Dec 3, 2010 at 11:01 PM
http://www.opensips.org/Resources/DocsCoreVar16#toc84
2010/12/7 Brett Nemeroff br...@nemeroff.com
Hello all,
I'm tring to get a full time string in the format:
12:24:19.000 CST Sat Dec 04 2010
Anyway I can do that without a hack?
Thanks,
Brett
Hi,
It depends on what will be the purpose of opensips (what functions you will
use, etc)
-Laszlo
2010/12/3 Vic Jolin victor.jo...@gmail.com
Hi all,
Just wanted to ask what is the hardware requirement to achieve at least
5000 cps? Is that even possible?
Thanks
auth.
-Laszlo
(It is just my experience, not a suggestion or a recommendation)
2010/12/3 Vic Jolin victor.jo...@gmail.com
Main purpose of the proxy will be for routing outbound and inbound calls.
Functions I might be using is authentication and lcr
On Fri, Dec 3, 2010 at 10:48 PM, Laszlo
These are real calls, or some kind of sipp generated traffic?
2010/12/3 Vic Jolin victor.jo...@gmail.com
When more than 100 calls, calls does not complete, it's like sip packets
are not being sent properly
2010/12/4 Stanisław Pitucha virap...@gmail.com
On 03/12/10 16:03, Vic Jolin wrote:
Ok, and no errors or something in the log when more than 100 calls, calls
does not complete, it's like sip packets are not being sent properly is
happening?
2010/12/3 Vic Jolin victor.jo...@gmail.com
Yes these are real calls
On Sat, Dec 4, 2010 at 12:55 AM, Laszlo las...@voipfreak.net wrote
post the SIP 300 captured by ngrep?
-Laszlo
Thanks,
Brett
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Sounds like a bad idea :)
You can try to play with the exec module, see
http://www.opensips.org/html/docs/modules/1.6.x/exec.html
-Laszlo
2010/7/24 Nauman Sulaiman nauman762-h...@yahoo.co.uk
Hi, using OpenSIPs 1.6, how does one run a custom php script (or whatever
code) upon
receiving
Brett is right, but avp_db_query can be a performance killer too (just
imagine it with 200 new calls per second = 200 insert queries/sec)
-Laszlo
2010/7/24 Nauman Sulaiman nauman762-h...@yahoo.co.uk
Thank you for the respnses
--- On *Sat, 24/7/10, Brett Nemeroff br...@nemeroff.com* wrote
You can relay the register to sipgate, so when auth is done, then you can
forward it.
consume_credentials();
$du = sip:sipgate_ip_goes_here:port;
t_relay();
exit;
so instead of doing if (!save(location)) etc
-Laszlo
2010/7/24 Nauman Sulaiman nauman762-h...@yahoo.co.uk
Hi, We have
Hi,
It doesn't accept a variable or an avp, just a clear text in
codec_exist();
The same is true for all codec_ operations.
-Laszlo
2010/7/14 Jozsef CZOMPO czompo.joz...@bitnet.hu:
Hi
I need to specify in database table the codec order per sip clients.
So i'm created a table with some
Also add loadmodule domain.so
2010/6/7 Premalatha Kuppan premala...@ngintech.com:
Hi,
Can someone tell me, how to recover from this error,
Jun 7 06:17:52 [10803] DBG:core:find_cmd_export_t: found sl_send_reply(2)
in module sl [//lib64/opensips/modules/]
Jun 7 06:17:52 [10803]
an one
size fits all module will work for LI.
-Laszlo
2010/5/24 Antonio Anderson Souza antonio...@gmail.com
RTPProxy is able to fork the RTPStream in Realtime to a new destination.
I guess it would be possible to implement a module (or enhance some one
existent, i think could be a B2B special
Server: OpenSIPS (1.5.1-notls (x86_64/linux))
Content-Length: 0
Hi,
Can you post the full trace where the whole message flow can be checked?
-Laszlo
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Did you try to search for the solution?
http://tinyurl.com/2f46nht
-Laszlo
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you have defined in opensips.cfg are listed
in disctionary.opensips.
But it's not an opensips error (and not an error in cdrtool).
-Laszlo
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-Laszlo
Before the INVITE would look like:
INVITE sip:+112355599...@1.1.1.1*;npdi=yes;rn=+1123555 SIP/2.0
but now it should look like INVITE
sip:+1123555;npdi=yes;rn=+112355599...@1.1.1.1* SIP/2.0
it currently looks like INVITE sip:+11235559...@*1.1.1.1* SIP/2.0.
Any ideals why
;
};
$du = $avp(s:destination);
$rd = $avp(s:reqdomain);
t_relay();
exit;
Not sure about the performance under a heavy inbound traffic (because of the
mysql queries).
-Laszlo
2010/5/1 Mike
Hi Ahmed,
As you can see, the other party gets local ip in SDP
c=IN IP4 192.168.0.168.
You can try to play with flags:
http://www.opensips.org/html/docs/modules/1.6.x/nathelper.html#id229028
-Laszlo
2010/4/30 Ahmed Munir ahmedmunir...@gmail.com
Hi.
Thanks for your reply, the traces
2010/4/29 David Adade david.ad...@dcomms.co.uk
Thanks
help much apreciated
modparam(rr, append_fromtag, 1)
modparam(uac,restore_mode,auto)
xlog($fu \n);
uac_replace_from(2101573...@10.10.1.100);
uac_replace_from(sip:2101573...@10.10.1.100 sip%3a2101573...@10.10.1.100
);
xlog($fu
itand follow the manual!
-Laszlo
2010/4/26 Adrian Georgescu [via OpenSIPS (Open SIP Server)] [hidden
email] http://user/SendEmail.jtp?type=nodenode=4961533i=0
On Apr 26, 2010, at 10:07 AM, samoh wrote:
Hi everybody,
I'm traying to install CDRTool since 3 days but no success
.
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Did you install 2 different versions of freeradius?
-Laszlo
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Users
lists.
It's outside the scope of this mailing list.
http://lists.opensips.org/pipermail/users/2010-April/012274.html
-Laszlo
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with the installation?
Which step was unsuccessful?
-Laszlo
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2010/4/22 samoh dahmani.s...@gmail.com
Hi Laszlo,
In my configuration I installed rtpproxy and freeradius but the CDRTool
don't works, I have this message in the web interface Cannot connect to
rating engine 192.168.0.190:9024 and in the network space Error:
mediaproxy
certificate file
and opensips?
Is there anything in the syslog/messages about this?
-Laszlo
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DNS to resolve the name of the acctserver?
(in radiusclient-ng/servers do you use IP or not?)
-Laszlo
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,
Chris
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Try to set mhomed=1
-Laszlo
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http
.
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Can you post your rows from opensips.cfg (starts with modparam(acc, so not
the whole file) ?
Did you also load aaa_radius?
-Laszlo
2010/4/22 Roberto Ovani roberto.e...@gmail.com
Il mercoledì 21/04/10 14.30, Laszlo ha scritto:
2010/4/21 Roberto Ovani roberto.e...@gmail.com
Il mercoledì 21/04/10 13.49, Laszlo ha scritto:
gcc -v
*The host with OPENSIPS :
*Using built-in specs.
Target: i486-linux-gnu
Configured
2010/4/22 Chris Maciejewski ch...@wima.co.uk
On 22 April 2010 17:30, Laszlo las...@voipfreak.net wrote:
2010/4/22 Chris Maciejewski ch...@wima.co.uk
Hi,
I'm trying to run two opensips instances accessing one MySQL database
using the usrloc db-only mode.
This results
2010/4/22 Chris Maciejewski ch...@wima.co.uk
On 22 April 2010 17:47, Laszlo las...@voipfreak.net wrote:
2010/4/22 Chris Maciejewski ch...@wima.co.uk
On 22 April 2010 17:30, Laszlo las...@voipfreak.net wrote:
2010/4/22 Chris Maciejewski ch...@wima.co.uk
Hi,
I'm trying to run two
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http://lists.opensips.org/pipermail/users/2008-November/001729.html
It can be a starting point.
-Laszlo
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2010/4/22 dcomms i...@dcomms.net
Hi,
I have configure the load balancer as per opensips
http://www.opensips.org/Resources/DocsTutLoadbalancing
My question is how can i forward registration requests to asterisk. What am
i doing wrong
Thanks
if (is_method(REGISTER)) {
#
2010/4/22 Laszlo las...@voipfreak.net
2010/4/22 dcomms i...@dcomms.net
Hi,
I have configure the load balancer as per opensips
http://www.opensips.org/Resources/DocsTutLoadbalancing
My question is how can i forward registration requests to asterisk. What
am
i doing wrong
Thanks
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