Toyima,
This question is asked many time before :). Kamailio and OpenSIPS have the same
roots. However, they differ more and more over time since each project is
going it's own way.
Bogdan has made a nice description of his (and the community's) vision on the
future of Opensips: http://www.ope
Hi,
I am using opensips 1.6.2.
I have modified the opensips.cfg script and it generates a memory leak.
The memory leak seems to be on pkmem since when I use the following command
'opensipsctl fifo get_statistics pkmem: shmem: | grep free', pkmem dicreases
I have followed the documenta
Hi Brett,
Brett Nemeroff wrote:
On Wed, Jan 12, 2011 at 9:04 AM, Bogdan-Andrei Iancu
mailto:bog...@voice-system.ro>> wrote:
Hi Gareth,
Gareth Blades wrote:
I am having a problem with running opensips in debug level 6.
When opensips is set to this I am finding that it
Hi Denis,
Be sure you the latest version of 1.6.4 (SVN/tarball/debs) - there was a
mem leak in the dialog module, issues that was fixed 1 or 2 weeks ago.
Regards,
bogdan
Denis Putyato wrote:
Hello!
I have such error messages in syslog while using 1.6.4
Jan 24 09:40:04 opensips
/us
Hi Ambert,
Just to be sure it is clear. When you configure a DB connection for a
certain module, OpenSIPS will use that particular connection to check in
a version table for the entries corresponding to that module.
So, for your case, opensips will check in the remote DB, in the version
tabl
Hi Jock,
Jock McKechnie wrote:
Greetings;
I apologise in advance for this one. I _know_ I screwed it up, but I
just cannot see how. I'm sure it's something blazingly obvious, but I
just cannot find it and it's driving me nuts.
I've written an OpenSIPS config that uses an external perl 'help
Hi Jesse,
Lost me a bit between those pieces of script.
Anyhow, as far I understood from your problem, you want to change the
FROM URI in order to use the new value later in the script (in the auth
part). Well, this does not work - in opensips, the changes you do over
the message are not visi
Hi Anton,
It seams that variables are not supported for this function params. More
or less as this functions are not so used in the late time (the idea of
using files for data provisioning is deprecated).
You have two options:
- use dialplan module to set rules for checking the allowance (
Hi Denis,
From SIP point of view, the BYE must be sent to the contact URIs . I
guess your contact is different than the layer3 IP because of some NAT
presence, right ? if so, use fix_nated_contact() for INVITE and 200 OK,
so that the received contact will be "fixed" with the layer3 IP, so the
Hi Dave,
do : if (src_ip==myself) {}
Regards,
Bogdan
Dave Singer wrote:
Is there any way to check if the source IP/port is one that opensips
is listening on or one ? something like if ("sip:$si:$sp" == myself) {
...bla; bla;}
Thanks
Dave
___
Users
Hi Guys,
Just to clarify a bit here - from OpenSIPS perspective, all LINUX
distro are fully supported (like compiling and runing) and most of
UNIX-like OS (BSD, SOLARIS, etc).
There is no difference is running OpenSIPS on Debian or RedHat - you
just need to take care of the dependencies (pa
Hi John,
Can I list you here:
http://www.opensips.org/Resources/Downloads#osipmi
Thanks and regards,
Bogdan
John Khvatov wrote:
Hello.
I maintain OpenSIPS package in fedora/epel.
Of course, let me know if you have issues with OpenSIPS rpm package from
official fedora/epel repos.
On 31.0
Hello Bogdan
" because of some NAT presence, right ?"
No, I need use IP address when there is more than one SIP proxy in call path.
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu
Sent: Wednesday, Febr
Hello,
regarding opensips-cp and drouting i came across a small problem, maybe
someone already tried something similar and wants to share his knowledge :)
|
opensips-cp --> Drouting / Settings, Gateway Types / Group ID?s is what
i am talking about.
|
Is there any function to check for the
Hello.
There is problem with $auth.resp script variable.
Line from opensips.cfg:
xlog("L_INFO", "$auth.resp\n");
opensips -f opensips.cfg -c results:
Feb 2 16:33:12 [24453] NOTICE:core:main: config file ok, exiting...
Runtime error:
Feb 2 16:34:27 aki /usr/sbin/opensips[24485]: ERROR:core:pv_
Hi Bogdan.
On 02.02.2011, at 15:45, Bogdan-Andrei Iancu wrote:
> Hi John,
>
> Can I list you here:
> http://www.opensips.org/Resources/Downloads#osipmi
Sure.
Also, you can update 'Fedora' (link: 'http://fedoraproject.org', latest version
of OpenSIPS available in all currently supported Fedor
Anyone doing Fedora RPMs for MediaProxy?
I have only a CentOS link here:
http://mediaproxy-ng.org/wiki/InstallationGuide
Adrian
On Feb 2, 2011, at 8:52 AM, John Khvatov wrote:
> Hi Bogdan.
>
> On 02.02.2011, at 15:45, Bogdan-Andrei Iancu wrote:
>> Hi John,
>>
>> Can I list you here:
>> htt
Hello, Adrian.
Unfortunately, MediaProxy is not available from Fedora/EPEL repositories. This
should be fixed :)
On 02.02.2011, at 16:55, Adrian Georgescu wrote:
> Anyone doing Fedora RPMs for MediaProxy?
>
> I have only a CentOS link here:
>
> http://mediaproxy-ng.org/wiki/InstallationGuide
Hi, using opensips 1.6.2. I am trying to use the user part of incoming RURI
to look up a contact, reason being is full RURI is incorrect, this is due to
bogus proxy upstream so need a workaround.
lookup("location") seems to be only if you use AOR.
For exmaple i need to reroute incoming ACK to r
Hi,
you could set OpenSIPS to not use domain part of uri, so your issue is
solved.
stefano
Il 02/02/2011 15:30, Nauman Sulaiman ha scritto:
Hi, using opensips 1.6.2. I am trying to use the user part of incoming RURI
to look up a contact, reason being is full RURI is incorrect, this is due to
Thanks for the answer, I was not very clear in my first email though,
sorry for that. Basically when caller A initiates a call, his $fu may
be pulled from the db and replaced with a uac_replace_from. Then if the
call progresses and is redirected to a new destination the call fails
to authentica
Hi Jesse,
in your script you are replacing from header twice.
Double check to your script and delete the second uac_replace_from.
This function can be used just once a phone call.
ciao
s
Il 02/02/2011 16:41, Jesse Cloutier ha scritto:
Thanks for the answer, I was not very clear in my first email
Hi.
Is it possible to log not to a 1 file via xlog?
WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru
___
Users mailing list
Users@lists.opensip
On Wed, Feb 2, 2011 at 10:10 AM, Anton Zagorskiy <
a.zagors...@oyster-telecom.ru> wrote:
> Hi.
>
> Is it possible to log not to a 1 file via xlog?
>
>
Not entirely sure what you are asking, but it's probably not an opensips
question. It's probably a syslog question. To log to more than one file,
y
Hello all,
Small problem with rtpproxy and nathelper
as you can see in the log bellow , I use the force_rtp_proxy method with
the 'r' flag to indicate to trust the IP of the SDP.
In the example bellow, the trusted IP is "95.128.80.6 3966" , the problem is
that the old rtp source (in th
Hi Guys,
I am testing the following call flow:
Soft Phone => opensips (configured for B2B) => third party termination SIP
proxy
Here is my config:
modparam("b2b_entities", "script_req_route", "b2b_request")
modparam("b2b_entities", "script_reply_route", "b2b_reply")
local_route {
xlog
On Wed, Feb 2, 2011 at 6:20 AM, Bogdan-Andrei Iancu wrote:
> Hi Jock,
>
>
> Jock McKechnie wrote:
>
>> Greetings;
>>
>> I apologise in advance for this one. I _know_ I screwed it up, but I just
>> cannot see how. I'm sure it's something blazingly obvious, but I just cannot
>> find it and it's driv
Hello,
Something is happening with my configuration, every time i restart opensips,
opensips lost the registrations and i have to restart the phonesby the
way, i configured the authentication and is not workink, for non-Register
and Register OpenSIPS is not sending the 401 or 407...i don't now
The B2B module is operating on the received INVITE. Any changes that
you make to the received INVITE are not visible by the B2B module.
Use a proxy to perform whatever you want to do (rtpproxy, accounting,
etc.) and a separate server only for b2b (top hiding).
Regards,
Ovidiu Sas
On Wed, Feb 2,
How did you started the session? Maybe you should start the session
from the beginning in "trusted" mode.
Regards,
Ovidiu Sas
On Wed, Feb 2, 2011 at 11:26 AM, Laurent Schweizer
wrote:
> Hello all,
>
>
>
> Small problem with rtpproxy and nathelper
>
>
>
> as you can see in the log bellow , I use
Hi Ovidu,
I do not perform any changes on the received invite.
The "top hiding" does it and the problem is.. it does not change only the
media IP. Everything else goes OK.
Are you saying the "top hiding" does not work properly with the nathelper ?
Thanks
-- Kamen
On 2 February 2011 19:41, O
The nathelper module is performing changes on the received INVITE
(changing the SDP).
Those changes are not visible by the b2b module and therefor discarded.
As a result, the nathelper module (and any module that is changing the
initial INVITE) doesn't work with the b2b module.
The only change visi
O ok thanks for the clarification. Just have to make sure to include
that in my web server code to make sure that table is always there
then.
On Wed, Feb 2, 2011 at 4:11 AM, Bogdan-Andrei Iancu-2 [via OpenSIPS
(Open SIP Server)]
wrote:
> Hi Ambert,
>
> Just to be sure it is clear. When you confi
Ok, that brings some clarification. Thanks.
So the correct scenario is:
1) softphone --registered to--> opensips A (pure)
2) call is relayed from opensips A to opensips B (the B2B one)
3) the opensips B connects to the termination
4) the RTP goes between the softphone -> opensips A -> rtpproxy
Wo
yes!
On Wed, Feb 2, 2011 at 1:12 PM, Kamen Petrov wrote:
> Ok, that brings some clarification. Thanks.
>
> So the correct scenario is:
> 1) softphone --registered to--> opensips A (pure)
> 2) call is relayed from opensips A to opensips B (the B2B one)
> 3) the opensips B connects to the terminati
Hey, I got some trouble with tls support for opensips, I'm using two blink
softphones, and i can't get them to communicate.
The communication between the client and the server is established, but when
i make calls, i got the a not found error:
here's my configuration's file :
Step 3: Create MySQL tables using the opensipsdbctl shell script. The syntax
for
this utility follows:
opensipsdbctl create
I'm getting the following error for the above syntax
bash: syntax error near unexpected token `newline'
what's wrong here ? might be silly question
_
for this case, it was started with the trusted flag just I have changed the
source.
Laurent
-Message d'origine-
De : Ovidiu Sas [mailto:o...@voipembedded.com]
Envoyé : mercredi 2 février 2011 18:45
À : OpenSIPS users mailling list
Objet : Re: [OpenSIPS-Users] Nathelper / rtpproxy , probl
Wow I missed that one. Thanks.
Does that work for PVs so I can test other IPs like one from another
header, say "X-src-ip:192.168.0.5". Last I tried I couldn't get it to
work. Not sure if that was 1.6.2 or 1.6.3. I'm using 1.6.4 now. :)
Thanks Again
Dave
On Wed, Feb 2, 2011 at 4:37 AM, Bogdan-And
I'm a flaming moron;
I had a local iptables in the way that I completely missed - I could see the
packets coming to the interface, so I assumed OpenSIPS must be "ignoring"
them - rather than it not getting them because iptables was blocking them.
I shall now crawl back to my hole in shame.
Thank
Anyone can help please in installing Opensip on Ubuntu.
On Wed, Feb 2, 2011 at 1:15 AM, Duane Larson wrote:
> The first thing you should do is
> http://www.packtpub.com/article/installation-of-opensips-1.6
>
> You can watch the webinars here
> http://www.opensips.org/Resources/We
The best place to start is http://www.opensips.org/
In the left column of the web page there is a section titled Resources
with links to many very helpful resources. Your using the mailing list
so you probably already have seen them to get here.
So. Where are you getting stuck? We need specifics in
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