Can you try the trunk version?
I remember fixing a bug related to the external IP, but I don't
remember if it is available on a stable version.
Regards,
Ovidiu sas
On Wed, Jul 6, 2011 at 11:16 AM, MingHon gming...@gmail.com wrote:
Hi Carsten,
I tried RTPProxy using external address. It work
You need to try the git trunk version.
The command is available on all versions, but it doesn't work.
Regards,
Ovidiu Sas
On Wed, Jul 6, 2011 at 10:01 PM, MingHon gming...@gmail.com wrote:
Hi Ovidiu,
Thanks for the info.
But why the command still available in the doc.
http://www.kamailio.org
Try the latest 3.1. If it doesn't work, install from git.
There should be instructions on the kamailio website aout how to do it.
Regards,
Ovidiu Sas
On Thu, Jul 7, 2011 at 1:47 AM, MingHon gming...@gmail.com wrote:
Hi,
thanks..
may i know which version of kamailio? what the different
3.1 was released on Oct 6, 2010 and my commit was on Oct 8, 2010:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commit;h=0af3944586c881f5d3ea0be19242cdf84906269e
You will need to wait for 3.2 or install from trunk.
Regards,
Ovidiu Sas
On Thu, Jul 7, 2011 at 1:49 AM, Ovidiu Sas o
Are you sure that force_rtp_proxy is invoked?
On Fri, Jul 1, 2011 at 11:00 PM, MingHon gming...@gmail.com wrote:
Hi list,
can anyone here help me out here?
i also tried putting it in reply_one.
but in the sdp the c= and o= did not change. it stil having the private ip.
and send PUBLISH out.
I would stick with asterisk handling the mwi subscription and
configure kamailio as a relay only for those events.
Regards,
Ovidiu Sas
On Thu, Jun 30, 2011 at 9:30 AM, Spinov Evgeniy
spinov_evge...@intalisan.com wrote:
This makes the main problem as in documents I've found
It is possible to use kamailio and rtpproxy behind a NAT.
You need to properly craft the config to deal with the fact that
kamailio is behind NAT (proper Via, Route, Record-Route headers) and
you need to do port forwarding for SIP signalling and rtp.
Regards,
Ovidiu Sas
On Wed, Jun 29, 2011
If you are using asterisk for voicemail, you should let asterisk
notify the subscribers about mwi.
If you want to use kamailio for notifications, then you need to
configure asterisk to send PUBLISH requests to kamailio and kamailio
will notify the subscribers.
Regards,
Ovidiu Sas
On Wed, Jun 29
).
And also you will need to put proper IPs in SDP (see the rtpproxy module).
Start with a simple config, make calls, check all the SIP requests,
find out what's not properly set and fix the config.
Regards,
Ovidiu Sas
On Wed, Jun 29, 2011 at 10:08 PM, MingHon gming...@gmail.com wrote:
Hi,
Can you give
:
https://sourceforge.net/tracker/?func=detailatid=743020aid=2158069group_id=139143
https://sourceforge.net/tracker/?func=detailatid=1086413aid=2166279group_id=232389
Regards,
Ovidiu Sas
On Thu, Jun 16, 2011 at 12:46 PM, Asgaroth 00asgarot...@gmail.com wrote:
Hi All,
I am trying to send a mi
IIRC, the feature was requested during openser project (before the
merge with sip-router) and it was lost after the merge.
You can open a new one on the sip-router tracker.
Regards,
Ovidiu Sas
On Thu, Jun 16, 2011 at 1:18 PM, Bruce McAlister
bruce.mcalis...@blueface.ie wrote:
Ahh yes, you
Take a look at the nathelper module:
http://www.kamailio.org/docs/modules/3.1.x/modules_k/nathelper.html#id2592146
Regards,
Ovidiu Sas
On Tue, Jun 14, 2011 at 6:52 PM, Lucas Alvarez luca...@gmail.com wrote:
Hi, I want to modify the following sip packet, I want to change the SDP IP
address
processed).
$avp may be transaction persistent (enable it in tm):
http://www.kamailio.org/dokuwiki/doku.php/pseudovariables:1.4.x#avps
http://www.kamailio.org/docs/modules/1.4.x/tm.html#id2530425
Regards,
Ovidiu Sas
On Wed, Jun 8, 2011 at 7:10 AM, Omar o...@321communications.com wrote:
Do i
Here's an example on how to perform forking on failure on 1.4:
https://openser.svn.sourceforge.net/svnroot/openser/branches/1.4/examples/serial_183.cfg
Regards,
Ovidiu Sas
On Tue, May 24, 2011 at 4:30 PM, Omar o...@321communications.com wrote:
I need to rewrite the destination after I receive
Hello Andrew,
A while ago I commit-ed an enhancement to the rr module to allow
setting two rr headers:
http://kamailio.org/docs/modules/devel/modules_k/rr.html#id2784744
Please test it and report back any issues.
Regards,
Ovidiu Sas
On Sun, May 22, 2011 at 9:40 AM, Andrew Pogrebennyk
, that was the expected behavior.
Not sure if this was fixed.
On 13/05/2011, Ovidiu Sas o...@voipembedded.com wrote:
Try the nat_traversal module:
http://www.kamailio.org/docs/modules/3.1.x/modules_k/nat_traversal.html
It should handle this kind of scenarios.
Regards,
Ovidiu Sas
On Fri
Just set the RURI to the value that you want and then call append_brach().
Everything that's in RURI will be appended as a new branch.
After that, reset the RURI to it's original value: revert_uri().
Regards,
Ovidiu Sas
On Sat, May 7, 2011 at 12:37 PM, alex pappas rebel.pap...@gmail.com wrote
LCR tables, of course:
http://www.kamailio.org/docs/modules/3.1.x/modules/lcr.html#id2551334
http://www.kamailio.org/docs/modules/3.1.x/modules/lcr.html#id2499833
http://www.kamailio.org/docs/modules/3.1.x/modules/lcr.html#id2501638
Regards,
Ovidiu Sas
On Wed, Apr 27, 2011 at 11:28 AM, Alida
The business mailing list is more appropriate for this kind of requests.
Here are some links that might help:
http://www.kamailio.org/w/business/
http://www.kamailio.org/w/business-directory/
Regards,
Ovidiu Sas
On Tue, Apr 5, 2011 at 9:28 AM, Sabatella, Michael
michael.sabate...@ipc.com wrote
if it ends up in a failure route, a flag can be set to identify it.
Regards,
Ovidiu Sas
On Thu, Mar 31, 2011 at 11:28 AM, Klaus Darilion
klaus.mailingli...@pernau.at wrote:
maybe you can use t_any_timeout? I do not know if it handles internal and
external timeout identical
http://www.kamailio.org
-location2 and so on ...
Anyway, you can't solve this issue with a single call from asterisk
unless you have some rtp-replicator somewhere in the middle to fork
the rtp stream from asterisk and let one single rtp stream to go back
to asterisk.
Regards,
Ovidiu Sas
to emulate an edge server with a well crafted config.
Regards,
Ovidiu Sas
On Wed, Mar 9, 2011 at 11:38 AM, Derrick Ding dd...@aastra.com wrote:
Dear All,
I wonder if Kamailio supports RFC5626 as edge proxy or outbound proxy.
That is,
Edge Proxy adds a Path header with a flow token
You need to check the dictionaries on your kamailio server.
Mos likely something is miss configured there.
Check what value do you have for User-Name and see if you have any
duplicates for that value.
Regards,
Ovidiu Sas
On Sat, Mar 5, 2011 at 2:32 AM, Kosilov Fedor dangerko...@gmail.com wrote
/kamailio.spec.CenOS;h=b2c6d013b2071ac7ec54c15c661b58e7e0d5b005;hb=HEAD
Regards,
Ovidiu Sas
On Wed, Mar 2, 2011 at 5:56 AM, Suresh Bhandari bring...@yahoo.com wrote:
Hello Ovidiu,
I searched through the kamailio.org/pub but can't find what you said.
Can you verify the exact location of package
Each module may or may not have statistics.
Check the README file for each module that you have loaded to see what
statistics are available.
For example, here are the statistics exported by the tmx module:
http://www.kamailio.org/docs/modules/3.1.x/modules_k/tmx.html#id2700650
Regards,
Ovidiu
There is an rpm spec file under ./pkg/kamailio/rpm/kamailio.spec.CenOS.
That one should work well under RHEL 5.
Regards,
Ovidiu Sas
On Tue, Feb 22, 2011 at 6:25 AM, Watkins, Bradley
bradley.watk...@compuware.com wrote:
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun
Add some logs (print the message that you are processing and the rtp
command that you are issuing).
That should help you in troubleshooting your scenario.
Regards,
Ovidiu Sas
On Thu, Feb 10, 2011 at 4:55 AM, Emil Kroymann emil.kroym...@isaco.de wrote:
Hi,
yeah, the script does call rtp_offer
from modules_k.
Regards,
Ovidiu Sas
On Thu, Feb 10, 2011 at 9:11 AM, Ovidiu Sas o...@voipembedded.com wrote:
But this is how the code it is!
Can you provide more details about which version of the code are you using?
And which modules?
Are you using the kamailio modules or the ser version
.
Regards,
Ovidiu Sas
On Thu, Feb 10, 2011 at 9:36 AM, Emil Kroymann emil.kroym...@isaco.de wrote:
Am Thu, 10 Feb 2011 09:23:51 -0500
schrieb Ovidiu Sas o...@voipembedded.com:
It seems that the s version of nathelper is the one that you are using
(I just checked the code and the bug is present
Please test the latest trunk version. I removed the rtpproxy
functionality from nathelper s version.
You can safely use now nathelper from s with rtpproxy from generic.
Regards,
Ovidiu Sas
On Thu, Feb 10, 2011 at 9:36 AM, Emil Kroymann emil.kroym...@isaco.de wrote:
Am Thu, 10 Feb 2011 09:23
rtpproxy module was moved to generic
rtpproxy functionality from nathelper (s) module was removed
Later on, we should merge the nathelper modules into a generic one.
Regards,
Ovidiu Sas
On Tue, Jan 18, 2011 at 1:49 PM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
Hello,
On 1/13/11 7:38
I will check that and I will get back to you.
Regards,
Ovidiu Sas
On Wed, Feb 9, 2011 at 11:48 AM, Emil Kroymann emil.kroym...@isaco.de wrote:
Hi,
We recently had a problem with the nathelper module and rtpproxy in a
scenario where the SDP offer is sent only in the 200 OK. We use
sip
#qis_it_possible_to_reload_openser_s_configuration_file_with_a_signal
Regards,
Ovidiu Sas
On Thu, Feb 3, 2011 at 1:38 PM, Stagg Shelton st...@vocalcloud.com wrote:
Is it possible to reload a module such as siptrace without requiring a full
restart of kamailio. I have changed my db_url modparam for the siptrace
module and would
) and (k) users:
- rtpproxy: a single module for dealing with rtpproxy servers;
- nathelper: two variants for dealing with NAT signaling.
Next step, will be to merge the two nathelper modules into a single one.
Thoughts?
Regards,
Ovidiu Sas
___
SIP
to be manually tweaked (if really necessary).
Regards,
Ovidiu Sas
But there is no similar feature for To header manipulation.
regards
klaus
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http
Run top for all processes and also run: vmstat 2
That will give you a global view of your system.
As you can see the proxy is not using a lot of cpu.
BTW, this is the wrong mailing list as it seems that you are running
opensips and not kamailio.
Regards,
Ovidiu Sas
On Thu, Dec 2, 2010 at 11
Try the textops module:
http://www.kamailio.org/docs/modules/3.1.x/modules_k/textops.html#id2907488
Regards,
Ovidiu Sas
On Wed, Nov 24, 2010 at 5:19 AM, Johannes Wagner-Meingassner
johannes.wagner-meingass...@infotech.at wrote:
Hi all!
I have to remove the Transparent Codec in the INVITE
in the record-route-header.
record_route_preset and record_route are exclusive - use one or the
other but not both for the same message
Regards,
Ovidiu Sas
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
mediaproxy and qos:
http://www.kamailio.org/docs/modules/3.1.x/modules/mediaproxy.html#id3080297
http://www.kamailio.org/docs/modules/3.1.x/modules_k/qos.html#id2747034
Regards,
Ovidiu Sas
On Wed, Oct 20, 2010 at 10:31 AM, Alex Balashov
abalas...@evaristesys.com wrote:
Not that I know
That's right. I forgot about that one:
http://www.kamailio.org/docs/modules/3.1.x/modules_k/sst.html#id2968725
Thanks,
Ovidiu
On Wed, Oct 20, 2010 at 11:25 AM, Timo Reimann timo.reim...@1und1.de wrote:
On 20.10.2010 16:47, Ovidiu Sas wrote:
mediaproxy and qos:
http://www.kamailio.org/docs
/examples/alg.cfg;hb=ad7f00d840082989132f335914aa0db223a0e46e
Regards,
Ovidiu Sas
On Fri, Oct 15, 2010 at 8:40 AM, Anders vae...@gmail.com wrote:
I've been looking around for documentation to understand rtp proxy
with Kamailio better, but with no luck (or I might be quite
thick-headed!).
Anyone know
Hello Joe,
Check out the example provided in the source tree:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob_plain;f=modules_k/rtpproxy/examples/4to6.cfg;hb=ad7f00d840082989132f335914aa0db223a0e46e
Regards,
Ovidiu Sas
On Thu, Oct 14, 2010 at 8:59 PM, Joe Uelk joe.u
that is carrying SDP and rtpproxy_answer for the subsequent reply or
ACK request that is carrying SDP. Take a look at the README file:
http://www.kamailio.org/docs/modules/3.1.x/modules_k/rtpproxy.html#id2759708
Regards,
Ovidiu Sas
On Fri, Oct 15, 2010 at 11:36 AM, Anders vae...@gmail.com wrote:
Thanks
a clean install.
Regards,
Ovidiu Sas
On Fri, Oct 15, 2010 at 12:29 PM, Anders vae...@gmail.com wrote:
Hi,
Sorry about the low-level question, but... - how do I upgrade from
Kamailio 3.0 to 3.1? I used git for installing 3.0 (this great guide:
http://www.kamailio.org/dokuwiki/doku.php
Regards,
Ovidiu Sas
On Thu, Oct 14, 2010 at 12:19 PM, Morten Isaksen mi...@misak.dk wrote:
Hi,
I was mistaken. This is not the problem. OCS kan handle r-r with multiple
entry.
This one works OK - The OCS sends a PRACK
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP x.x.42.177:65371;rport=65371
) );
Regards,
Ovidiu Sas
On Thu, Oct 14, 2010 at 12:22 PM, Nathan Angelacos nan...@nothome.org wrote:
Hi Nicolas,
On 10/14/10 08:50, Nicolas Rüger wrote:
Hello,
I'm trying to get dialog module working..
...my problem is that nothing is stored in the database in table dialog
yet. I'm looking
dialog
or for requests that are already part of an existing dialog is useless
(it's a noop).
Regards,
Ovidiu Sas
On Thu, Oct 14, 2010 at 12:22 PM, Nathan Angelacos nan...@nothome.org wrote:
Actually, you want dlg_manage() to be called for every message, not just
INVITES
route
/alg.cfg;hb=HEAD
Regards,
Ovidiu Sas
On Wed, Oct 6, 2010 at 6:41 PM, Stagg Shelton st...@vocalcloud.com wrote:
I'm working on a setup where we have rtpproxy on a machine with eth0 IP
10.10.5.141/19 and eth1 IP 10.10.10.78/24. When using
force_rtp_proxy(,10.10.5.141); the connection information (c
They work ok in my setup. The plan is to remove force_rtp_proxy in 3.2.
Regards,
Ovidiu Sas
On Wed, Oct 6, 2010 at 7:19 PM, Alex Balashov abalas...@evaristesys.com wrote:
I assume that means rtpproxy_answer|offer actually work now. :-)
Last time there was a conversation about using them
work properly now (as opposed to
previous versions 3.1).
Regards,
Ovidiu Sas
On Wed, Oct 6, 2010 at 7:24 PM, Alex Balashov abalas...@evaristesys.com wrote:
On 10/06/2010 07:22 PM, Ovidiu Sas wrote:
They work ok in my setup. The plan is to remove force_rtp_proxy in
3.2.
A wholeheartedly
Hello Juha,
The awk version of the script is now integrated in kamctl.
Regards,
Ovidiu Sas
On Wed, Sep 29, 2010 at 6:59 AM, Juha Heinanen j...@tutpro.com wrote:
Daniel-Constantin Mierla writes:
just to mention that kamctl base depends on awk, so this language can
be used in case
On Tue, Sep 28, 2010 at 11:10 AM, Andrei Pelinescu-Onciul
and...@iptel.org wrote:
On Sep 28, 2010 at 18:04, Rouskol Andrey anry-...@yandex.ru wrote:
Andrei,
When I recompile kamailio with debug options you specified, it works just
fine! (no memdbg and memlog options in cfg file). If I
Daniel,
What do you think about moving the (k) presence modules to standard
for the next 3.1 kamailio release?
Regards,
Ovidiu Sas
On Tue, Sep 28, 2010 at 11:38 AM, Andrei Pelinescu-Onciul
and...@iptel.org wrote:
On Sep 28, 2010 at 11:29, Ovidiu Sas o...@voipembedded.com wrote:
On Tue, Sep
If the script can be provided in bash, then it would be nice to be
integrated in kamctl:
- kamctl lcr weight_eval weight_1 weight_2 ... weight_n
Regards,
Ovidiu Sas
On Mon, Sep 27, 2010 at 1:53 PM, Juha Heinanen j...@tutpro.com wrote:
Daniel-Constantin Mierla writes:
you can create as well
On Wed, Sep 22, 2010 at 3:42 AM, Klaus Darilion
klaus.mailingli...@pernau.at wrote:
Am 21.09.2010 17:12, schrieb Ovidiu Sas:
Hello all,
I would like to propose the removal of force_rtp_proxy function from
the rtpproxy (k) module.
Instead, the rtpproxy_offer/rtpproxy_answer should be used
of force_rtp_proxy is provided by
rtpproxy_offer/rtpproxy_answer;
- rtpproxy_offer/rtpproxy_answer is easier to use in the config.
Objections?
Regards,
Ovidiu Sas
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip
/answer is just extra work load
(given the fact that both are just wrappers around a single function).
I would go for a clean layout ... but ... Vox Populi, Vox Dei :)
Regards,
Ovidiu Sas
On Tue, Sep 21, 2010 at 11:32 AM, Alex Balashov
abalas...@evaristesys.com wrote:
On 09/21/2010 11:27 AM, Daniel
The kamailio doc website doesn't have the new rtpproxy module documentation:
http://www.kamailio.org/docs/modules/devel/
Also, ratelimit module is listed as being k and in fact it is generic now.
Regards,
Ovidiu Sas
___
SIP Express Router (SER
Long time ago I did a brief description on how bridging can be
achieved. It was for openser but it is still valid:
http://www.mail-archive.com/us...@openser.org/msg04806.html
Probably we should add this to the rtpproxy module documentation.
Regards,
Ovidiu Sas
On Tue, Sep 21, 2010 at 12:32 PM
There is no other more user friendly procedure for the 200ok/ACK SDP
negotiation.
The existence of SDP in the initial INVITE must be checked.
Regards,
Ovidiu Sas
On Tue, Sep 21, 2010 at 8:16 PM, Iñaki Baz Castillo i...@aliax.net wrote:
2010/9/21 Ovidiu Sas o...@voipembedded.com:
Why
and analyze the jitter on both incoming and outgoing rtp stream.
If this doesn't have any effect, then the code needs to be
instrumented to see what exactly is going on (by adding specific
probes for your specific tests).
It would be interesting to see your results.
Regards,
Ovidiu Sas
On Sat, Sep
.
Regards,
Ovidiu Sas
On Fri, Sep 17, 2010 at 8:58 PM, Pranav Desai pde...@signalogic.com wrote:
Hello,
We have a proxy server setup that runs Kamailiov3.0.1 + RTPProxyv1.2.1.
Nathelper support has been enabled in Kamailio.
On a certain route, we call force_rtp_proxy(z120), i.e
Radius can be enabled without modifying the Makefile. Before
compiling set ENABLE_RADIUS_ACC=true
ENABLE_RADIUS_ACC=true make
ENABLE_RADIUS_ACC=true make install
Regards,
Ovidiu Sas
On Wed, Sep 8, 2010 at 2:24 PM, dotnetdub dotnet...@gmail.com wrote:
On 8 September 2010 15:00, Ovidiu Sas o
://www.kamailio.org/docs/modules/3.0.x/modules_k/nathelper.html#id2610143
After that, before invoking force/unforce_rtp_proxy, you will need to
set the proper set id for each rtpproxy server:
http://www.kamailio.org/docs/modules/3.0.x/modules_k/nathelper.html#id2607313
Regards,
Ovidiu Sas
On Thu, Sep
/ obsolete
git mv module_k/ratelimit modules
Regards,
Ovidiu Sas
On Tue, Aug 31, 2010 at 5:29 PM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
Hello,
On 8/31/10 6:24 PM, Ovidiu Sas wrote:
Hello all,
The (k) ratelimit module is an enhanced version of the original (s)
ratelimit module
,
Ovidiu Sas
On Thu, Jun 24, 2010 at 9:42 AM, Stagg Shelton st...@vocalcloud.com wrote:
I have a customer who has just started operating a dialer. Over the past
couple of days this dialer has been creating some minor issues with our US48
termination. We are currently running Server:: Kamailio
On Wed, Jun 23, 2010 at 9:25 AM, Timo Reimann timo.reim...@1und1.de wrote:
Hi,
Ovidiu Sas wrote:
On Tue, Jun 22, 2010 at 12:27 PM, Timo Reimann timo.reim...@1und1.de wrote:
2.) Various race conditions may generate log messages such as following
when DID mode is enabled
On Tue, Jun 22, 2010 at 12:27 PM, Timo Reimann timo.reim...@1und1.de wrote:
Hello,
I have two issues related to dialog module state keeping/logging that
caught my attention today:
1.) The module considers requests routed in early (but not yet
confirmed) dialogs to be bogus, as can be seen
Now, that you got everything you need, you should document this on the
wiki (so others could find the info without asking on the mailing
list).
Regards,
Ovidiu Sas
On Wed, May 5, 2010 at 7:08 PM, dotnetdub dotnet...@gmail.com wrote:
Thanks Alex.. Excellent information.
Regards,
Stephen.
On 5
No AVPs or vars. The encoding field separator can be specified via params:
http://sip-router.org/docbook/sip-router/branch/master/modules_k/siputils/siputils.html#id2574949
Regards,
Ovidiu Sas
On Tue, May 4, 2010 at 9:41 AM, Uriel Rozenbaum
uriel.rozenb...@gmail.com wrote:
Thanks Guys, I'll
in-dialog request, you check if the RURI is encoded
and you decode and then route normally.
http://kamailio.org/docs/modules/stable/modules_k/siputils.html#id2587362
I was using this method with success with several switches.
Hope this helps.
Regards,
Ovidiu Sas
On Fri, Apr 30, 2010 at 4:10 PM, Uriel
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