db on a separate machine.
Regards,
Ovidiu Sas
On Wed, Aug 26, 2009 at 4:21 PM, Aryanto Rachmad wrote:
> Hello Bogdan,
>
> My question is actually related to one of the points listed on the
> feature list, which said:
>
> "OpenSIPS can run on embedded systems, with limited re
x/qos.html#id227316
Regards,
Ovidiu Sas
On Tue, Sep 1, 2009 at 9:29 AM, urmi lakkad wrote:
> Hello,
>
> I am using Opensips-1.5.1 with Asterisk.
> I want to test the QOS module functionality. I have configured the dialog
> module and its working fine.
> Even I have set the flag of
That is the exact purpose of ratelimit module.
You can do automatic ratelimit as defined in the params or you can do
forced ratelimiting for every new INVITE (which will trigger a new
call).
It is very powerful and flexible.
Regards,
Ovidiu Sas
On Thu, Sep 3, 2009 at 4:01 PM, Brett Nemeroff
Hello Brett,
See inlines.
Regards,
Ovidiu Sas
On Thu, Sep 3, 2009 at 9:37 PM, Brett Nemeroff wrote:
> I'm trying to wrap my head around this conceptually. I don't understand the
> forced ratelimiting you mention. Where is that in the docs?
http://www.opensips.org/html/do
For the b2b module, I think it will be a good idea to have a parameter
that controls the tear down of the call when a timeout occurs
(disconnecting the two legs by sending BYEs in both directions, just
like the dialog module).
Regards,
Ovidiu Sas
On Fri, Oct 9, 2009 at 3:42 AM, Bogdan-Andrei
Session timers it's tricky. It needs to be properly implemented by
both refresher and refreshee.
The issue here is that if you have a call in b2b mode and the b2b call
times out on the opensips server, all subsequent in dialog message
will be discarded by the opensips server.
Regards,
Ovidi
The doc was incomplete. This feature was already implemented for the b2b module.
The module README was updated to reflect this:
http://www.opensips.org/html/docs/modules/devel/b2b_logic.html#id227308
In the end, it was a "false alarm" :)
Regards,
Ovidiu Sas
On Fri, Oct 9, 2009 a
It is a simple timeout.
Regards,
Ovidiu Sas
On Fri, Oct 9, 2009 at 1:40 PM, Saúl Ibarra wrote:
> On Fri, Oct 9, 2009 at 7:36 PM, Ovidiu Sas wrote:
>> The doc was incomplete. This feature was already implemented for the b2b
>> module.
>> The module README was updated to
aware are the maximum number of pipes:
http://www.opensips.org/html/docs/modules/devel/ratelimit.html#id271306
If you need more then 16 pipes, you will need to recompile.
Hope this helps.
Regards,
Ovidiu Sas
On Mon, Nov 9, 2009 at 11:31 AM, Jeff Pyle wrote:
> Hello,
>
> It appears our opti
Hello Daniel,
You can post your patch on the rtpproxy devel mailing list:
http://lists.rtpproxy.org/mailman/listinfo/devel
Regards,
Ovidiu Sas
On Thu, Nov 19, 2009 at 3:37 PM, Daniel Goepp wrote:
> We did a little custom work to rtp proxy to support putting it behind NAT.
> We have been
Check the docs:
http://www.opensips.org/html/docs/modules/devel/nathelper.html#id271367
- flag 'c'.
Two "c=" lines are ok according to the rfc.
Regards,
Ovidiu Sas
On Thu, Nov 19, 2009 at 4:19 PM, Iñaki Baz Castillo wrote:
> Hi, I receive a wrong SDP from a gateway:
&g
rg/html/docs/modules/1.6.x/dispatcher.html#id271244
We could use algorithm id '8' for 'first entry is chosen'. This will
be totally backward compatible with 1.6.0.
Regards,
Ovidiu Sas
On Thu, Dec 17, 2009 at 10:25 AM, Bogdan-Andrei Iancu
wrote:
> Hi all,
>
> before
ot; instead of being whatever number ?
>
> Regards,
> Bogdan
>
> Ovidiu Sas wrote:
>> Hello Bogdan,
>>
>> I came across this issue with the dispatcher module: I need to
>> dispatch calls using the 'non implemented' method - the first entry in
>>
Yes, you can use the ratelimit module to control the cps.
You will need to assign a pipe for each outbound destination.
Regards,
Ovidiu Sas
On Mon, Feb 15, 2010 at 11:50 AM, rajib deka wrote:
> Hi Bogdan,
>
> I agree with you. But I have seen that RATELIMIT module is doing something
&g
0")
=> 200/5=40cps
In this case, you can have 200 messages in the first second and no
messages for the next 4s and the cps limitation will not kick in.
For the first second you will have a cps of 200 and for the next 4s
the cps will be 0, but on average you will have 40cps.
Hope this help
You need to create a dialog for calls between users involved in dialog
presence *and* you will also need mark the dialogs that will generate
PUBLISH: dialoginfo_set().
See:
http://www.opensips.org/html/docs/modules/1.6.x/pua_dialoginfo.html#id228322
Regards,
Ovidiu Sas
On Tue, Mar 2, 2010 at 1
ave authorization credentials.
Hope this helps. Next time, capture with timestamp info (-t option for ngrep).
Regards,
Ovidiu Sas
On Sat, Apr 24, 2010 at 9:59 AM, Andy Thomas wrote:
> Hi
>
> Im running 1.6.2 no TLS
>
> Config file is the chapter 5 demo file from Flavio’s book.
>
>
As I pointed out in a previous e-mail, the issue is with your client:
it should send out the second REGISTER with an Authorization header.
Please read the answers that you get before re-posting the same thing
again and again.
Regards,
Ovidiu Sas
On Tue, Apr 27, 2010 at 2:47 PM, Andy Thomas
/html/docs/modules/devel/auth.html#id228317
Maybe this will make your SIP UA happy.
Regards,
Ovidiu Sas
On Tue, Apr 27, 2010 at 3:54 PM, Andy Thomas wrote:
> I never got your first email.
> It wasn’t in the digest either!
>
> I have tried several sip client softphones, including x-l
Please keep the mailing list in copy. Private e-mails will go unanswered.
Can you post a trace of an X-lite trying to register?
Regards,
Ovidiu Sas
On Wed, Apr 28, 2010 at 2:42 PM, Andy Thomas wrote:
> I AM experiencing the exact same issue with Xlite
> And also if I try and regis
iving with clients that
are not sending from tag).
Regards,
Ovidiu Sas
On Tue, Dec 16, 2008 at 10:51 AM, Bogdan-Andrei Iancu
wrote:
> Hi,
>
> it seams that the reply has no Contact header (see the error), so no
> contact is stored into the dialog. Most probably the module tries later
>
dialog pov). The dialog module is already complex enough and if we
add more complexity for dealing with extremely broken clients doesn't
make sense.
I wonder if the broken 200ok was generated by a real UAS or a sipp script.
Regards,
Ovidiu Sas
On Tue, Dec 16, 2008 at 11:34 AM, Bogdan-Andr
Can you get a backtrace? I'm not aware of any issues related to the
ratelimit module.
Next time, please post your questions to the relevant mailing list.
Regards,
Ovidiu Sas
On Mon, Dec 29, 2008 at 3:37 PM, Bogdanov Roman wrote:
> Good evening.
>
> I recently compiled rateli
Hello Jeff,
Take a look also at the qos module. It keeps per dialog track of the
negotiated sdp.
There's an API that can be used by other modules sitting on top of the
qos module.
Regards,
Ovidiu Sas
On Thu, Jan 29, 2009 at 8:20 AM, Jeff Pyle wrote:
> Hi Bogdan,
>
> I
If you mix up ratedecks from at least three different NA carriers
(NPA/NXX/B) you can easily end up with 1.5 mil routes ...
Regards,
Ovidiu Sas
On Wed, Feb 18, 2009 at 9:49 AM, Brett Nemeroff wrote:
> I am curious, tho, why it's so big. I've done large, multi-carrier LCR
>
roxy when handling with
> SessionTimers.
>
> As I already said before, SST only works if caller and/or callee supports it
> (and uses it).
That's the role of the sst module: to enforce sst if one of the
participants supports it.
Regards,
Ovidiu Sas
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Or in 1.5 using the new T_rpl pv:
http://www.kamailio.net/dokuwiki/doku.php/pseudovariables:devel#t_rpl_pv
In your example: $T_rpl($ct) (inside the failure route).
Regards,
Ovidiu Sas
On Fri, Mar 13, 2009 at 5:06 PM, Iñaki Baz Castillo wrote:
> El Viernes, 13 de Marzo de 2009, Jeff P
Ooops, this is an opensips mailing list.
Sorry for misleading (to many mailing lists).
Inaki's solution will work ok.
Regards,
Ovidiu Sas
On Fri, Mar 13, 2009 at 5:36 PM, Ovidiu Sas wrote:
> Or in 1.5 using the new T_rpl pv:
> http://www.kamailio.net/dokuwiki/doku.php/pseudovari
Or browsing the opensips website:
http://www.opensips.org/index.php?n=Main.News0010
Regards,
Ovidiu Sas
On Sat, Mar 14, 2009 at 1:50 PM, Iñaki Baz Castillo wrote:
> El Sábado, 14 de Marzo de 2009, Marcus Vinicius escribió:
>> Hi,
>>
>> Someone could recommend me so
It seems that you are trying to create a dialog while processing a BYE.
The way dialog module works has changed between 1.4 and 1.5.
Check your script and make sure that you don't create a dialog for
messages other then initial INVITE.
Regards,
Ovidiu Sas
On Mon, Mar 16, 2009 at 10:26 PM,
s.org/html/docs/modules/1.4.x/lcr.html#id271156
Regards,
Ovidiu Sas
On Thu, Mar 19, 2009 at 4:46 PM, Noel R. Morais wrote:
> Hi Guys,
>
> is there a way to use a regex as prefix? I know, prefix is prefix, but
> for my routing rules it will be hard to build all the routes without
>
1.4.x/mi_xmlrpc.html
Regards,
Ovidiu Sas
On Fri, Mar 20, 2009 at 1:46 PM, Jeff Pyle wrote:
> Hello,
>
> Is it possible to have mi_datagram listen on more than one socket? For
> example, I need it to listen to a local socket for communication with
> Mediaproxy¹s dispatcher, but I¹d
The dialplan module may help here in matching those prefixes and
identifying the carrier.
And this will be faster then performing db lookups and maybe more
elegant then using the cache.
Regards,
Ovidiu Sas
On Thu, Mar 26, 2009 at 8:45 AM, Bogdan-Andrei Iancu
wrote:
> hi Brett,
>
> wel
You should be able to do all this with stock opensips and a lot of
transformations:
http://opensips.org/index.php?n=Resources.DocsCoreTran
Cordialement,
Ovidiu Sas
On Tue, Mar 31, 2009 at 2:36 PM, Francois Menard wrote:
> Folks,
>
> We have recently become a CLEC and we are entitled
In this case the CRITICAL should be turned into a NOTICE or WARNING.
And maybe a little more verbosity (type of reply ignored for
transaction with Call-ID) so it can be correlate with captured SIP
traffic.
Regards,
Ovidiu Sas
On Fri, Apr 3, 2009 at 12:33 PM, Bogdan-Andrei Iancu
wrote:
>
carrierroute will be deprecated in opensips.
opensips 1.5 carrierroute is identical with kamailio carrierroute 1.4
If you plan to use opensips, you should take a look at drouting module.
If you want to stick with the latest carrierroute, then you will need
to use kamailio.
Hope this helps,
Ovidiu
On Thu, Apr 23, 2009 at 7:58 AM, Ricardo Carvalho
wrote:
> It seems like I was doing everithing right, capturing the call sequence with
> wireshark revealed that a new branch is in fact created, but because this
> branch keeps the same Call-ID of the initial ENUM call, the second INVITE
> sent to
Open a bug in the tracker. Drouting should accept NULL values as
prefix and treat them as empty strings.
Regards,
Ovidiu Sas
On Thu, Apr 30, 2009 at 5:54 PM, Noel R. Morais wrote:
> Hi Guys,
>
> I would like to know How do I set a default rule in drouting module using
> Oracle? Th
.
If something is unclear on the doc file, let me know and I will improve it.
Regards,
Ovidiu Sas
On Thu, May 7, 2009 at 3:33 PM, k1028 wrote:
>
> I tried to google around and look through all the documentation but I am
> still unclear how the QoS module work to test it on my stagin
signaling
point of view, we don't know which codec will be used from a list of
multiple codec offer.
The qos module is keeping track of negotiated sdp sessions via
INVITE/200ok, 200ok/ACK. Support for UPDATE is present but not
implemented.
Support for early media is available.
Regards,
Ovidi
The core has an SDP parser which is able to parse and correlate codec
names with payload types. The only module using the sdp parser is the
qos.
The parsed sdp structure can be inspected via dialog mi commands:
http://lists.opensips.org/pipermail/devel/2008-December/001708.html
Regards,
Ovidiu
Hello Thomas,
The qos module keeps it's data inside the module.
Maybe later on, we will add to the dialog module a generic way of
storing data for modules sitting on top of the dialog module.
But for now, on restart, the qos context is lost.
Regards,
Ovidiu Sas
On Tue, Jun 23, 2009 at 1:
module functionality.
Regards,
Ovidiu Sas
On Wed, Jun 24, 2009 at 5:42 AM, Bogdan-Andrei
Iancu wrote:
> Hi Ovidiu,
>
> But the dialog module does provide a way of storing per dialog values.
> Indeed, this is available only from script, but it can be easily provided
> via the inter
Hello Andrei,
Are you using the core sdp parser implementation?
Ovidiu
On Wed, Jul 22, 2009 at 8:59 AM, andrei dragus wrote:
>
>
> Hi Jeff,
>
> This is Andrei. I am currently working on this right now. It should be
> finished in a couple of days if nothing comes up.
>
> Andrei.
>
> --- On Wed,
Perfect. Let me know if you need more features from the sdp parser.
Thanks,
Ovidiu
On Wed, Jul 22, 2009 at 11:32 AM, andrei dragus wrote:
>
> Yes.
>
> --- On Wed, 7/22/09, Ovidiu Sas wrote:
>
>> From: Ovidiu Sas
>
>> Hello Andrei,
>>
>> Are yo
Try 'make realclean' before a 'make install'. This should clean up
completely the repo.
Regards,
Ovidiu Sas
On Thu, May 27, 2010 at 11:45 AM, Richard Revels wrote:
> Yep. Did a make clean and make all before make install. I think its a
> little deeper than that.
&g
I don't think that the second boundary looks ok:
Boundary: \r\n--1fZ25o5K6UJOKcK25Mw2VZNJ\r\n
See the \r\n in the middle of the string.
Regards,
Ovidiu Sas
On Mon, May 31, 2010 at 2:39 PM, Juha Heinanen wrote:
> Adrian Georgescu writes:
>
>> Is also possible that being UDP,
Long time ago I tested rtpproxy with asymmetric clients and it worked ok.
IIRC, the 'a' flag must be used while forcing RTP packets through the rtpproxy.
Regards,
Ovidiu Sas
On Thu, Jun 17, 2010 at 1:41 PM, Adrian Georgescu wrote:
> We dropped support for asymmetric routing in M
.. for the given grp ip mask port
Regards,
Ovidiu Sas
On Fri, Aug 20, 2010 at 11:37 AM, logan wrote:
> Figured it out.
>
> 'opensipsctl address show' DOES read the DB
>
> 'opensipsctl address dump' dumps the addresses in memory
&g
The nathelper code is checking if the request for which rtpproxy_offer
is invoked is INVITE.
Just add UPDATE along the INVITE check and it should work fine (see
the attached patch).
Regards,
Ovidiu Sas
On Fri, Sep 17, 2010 at 5:59 PM, Daniel Goepp wrote:
> Is there support for rtpproxy_of
Hello Bogdan,
You are right. I will shortly remove both checks leaving the force
and offer/answer invocation up to the script writer.
Regards,
Ovidiu Sas
On Tue, Sep 21, 2010 at 5:32 AM, Bogdan-Andrei Iancu
wrote:
> Hi Ovidiu,
>
> IMHO, it is a bit of none-sense to have the the
By inspecting the code, I would suggest removing the old
force_rtp_proxy along with the 's' flag (which is already deprecated).
Keeping only rtpproxy_offer/rtpproxy_answer provides a cleaner design.
Comments?
Regards,
Ovidiu Sas
On Tue, Sep 21, 2010 at 8:28 AM, Ovidiu Sas wrote:
>
_rtp_proxy is provided by
rtpproxy_offer/rtpproxy_answer;
- rtpproxy_offer/rtpproxy_answer is easier to use in the config.
Objections?
Regards,
Ovidiu Sas
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Should I understand that all rtpproxy users have switched to
rtpproxy_offer/rtpproxy_answer?
If there are no objections, sometimes next week I will remove the old
force_rtp_proxy function.
Regards,
Ovidiu Sas
On Tue, Sep 21, 2010 at 10:57 AM, Ovidiu Sas wrote:
> Hello all,
>
> I woul
Excellent. We are making progress here. Later on, we will need to
break the nathelper module in two:
- nathelper: dealing with NAT related issues
- rtpproxy - connector to the rtpproxy server
Regards,
Ovidiu Sas
On Thu, Sep 23, 2010 at 3:30 AM, Bogdan-Andrei Iancu
wrote:
> Ovi
I suggest to remove those functions from the lcr code for the next
release and update the README file.
Regards,
Ovidiu Sas
On Tue, Sep 28, 2010 at 11:14 AM, Bogdan-Andrei Iancu
wrote:
> Hi Taiso,
>
> the load_contacts() and next_contact() are deprecated, better use the
> co
I still think that there are some users still using the lcr module ...
I will open a new thread about this and if there are no objections, we
will remove the lcr module for the next release.
Regards,
Ovidiu Sas
On Tue, Sep 28, 2010 at 11:42 AM, Bogdan-Andrei Iancu
wrote:
> Actually I wo
the next release.
Why:
- lcr module was deprecated by drouting module
- lcr module is no longer actively maintained
If there are any objections, please let us know.
Regards,
Ovidiu Sas
___
Users mailing list
Users@lists.opensips.org
http
If there are no objections, I would propose the removal of the module
sometimes next week.
Regards,
Ovidiu Sas
On Tue, Sep 28, 2010 at 12:03 PM, Ovidiu Sas wrote:
> Hello all,
>
> Following a different thread:
> http://lists.opensips.org/pipermail/users/2010-September/014713
You need to load both b2b_entities and b2b_logic:
http://www.opensips.org/html/docs/modules/1.6.x/b2b_logic.html#id228233
Regards,
Ovidiu Sas
On Wed, Oct 6, 2010 at 12:10 PM, David Santiago
wrote:
> Hello,
>
> I'm trying to test the "top hiding" scenario but I get
extra header(s) are available,
then those header(s) will be appended to the request.
Regards,
Ovidiu Sas
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Check out the documentation:
http://www.opensips.org/Resources/DocsCoreVar16#toc88
Read the docs, many questions are answered there ;)
Regards,
Ovidiu Sas
On Fri, Oct 15, 2010 at 12:22 PM, Daniel Goepp wrote:
> I see several functions for searching, removing, substituting or appending
&
Do you have any extra code in your repo? Or do you have a modified Makefile?
Trunk compiles just fine for me.
Regards,
Ovidiu Sas
On Tue, Oct 26, 2010 at 12:26 PM, Brett Nemeroff wrote:
> Bogdan,
> I must be doing something wrong. I just updated to 7329 and I'm
> getting the same
opensipsdbctl should be able to properly create the b2b tables in 1.6
now (I defined them in the list of EXTRA_MODULES).
Regards,
Ovidiu Sas
On Thu, Oct 28, 2010 at 5:39 PM, Brett Woollum wrote:
> Got it.
>
> I've updated the MySQL table and started OpenSIPS (which is version 1.6
If sip_reply_route is enabled, then manual accounting should be possible:
http://www.opensips.org/html/docs/modules/1.6.x/b2b_entities.html#id250001
http://www.opensips.org/html/docs/modules/1.6.x/acc.html#id293998
Regards,
Ovidiu Sas
On Tue, Nov 2, 2010 at 12:56 PM, Anca Vamanu wrote:
> On
You can do manual accounting:
http://www.opensips.org/html/docs/modules/1.6.x/acc.html#id294003
Or, you can create a new transaction, flag it for acc and then
terminate it t_reply:
http://www.opensips.org/html/docs/modules/1.6.x/tm.html#id293687
Regards,
Ovidiu Sas
On Tue, Nov 16, 2010 at 12
It seems that you will need to stick to manual accounting.
Regards,
Ovidiu Sas
On Tue, Nov 16, 2010 at 1:02 AM, Denis Putyato wrote:
> Thank you for reply
>
> First variant is not quite flexible for me.
> The second variant more interesting, but it doesn't work
>
&g
Deal with the host and port via PVs:
http://www.opensips.org/Resources/DocsCoreVar16#toc59
http://www.opensips.org/Resources/DocsCoreVar16#toc64
Regards,
Ovidiu Sas
On Wed, Nov 17, 2010 at 10:58 AM, Anton Zagorskiy
wrote:
> Hello.
> How to pass a value to the rewritehost() function usi
make sure that that all your SIP messages
are properly formated.
Because you are bridging traffic from/to your private network to/from
the public domain, make sure that signaling is properly formated for
both sides (private and public).
Regards,
Ovidiu Sas
On Sun, Nov 21, 2010 at 4:17 PM, aleks
You will need to identify the scenario type for each call and then
properly craft the INVITE.
There's no such thing as "local network" in opensips.
Regards,
Ovidiu Sas
On Fri, Nov 26, 2010 at 7:10 PM, Stefano Pisani
wrote:
> Hallo,
> my architecture needs the Opensips b
Please do not send private e-mails (keep the mailing list on cc).
Future e-mails will go unanswered.
You can play with the pemissions module to define and check against a
specific network.
http://www.opensips.org/html/docs/modules/1.6.x/permissions.html
Regards,
Ovidiu Sas
On Sun, Nov 28, 2010
hiding while the existing
proxy server will give you accounting.
Regards,
Ovidiu Sas
On Fri, Dec 3, 2010 at 10:48 AM, Jeff Pyle wrote:
> Hello,
> I am having trouble understanding exactly how the top-hiding scenario does a
> call compared to one that simply uses the tm module to relay it. B
erver with manual radius acc.
There are some issues with logging fake replies in b2b reply route
(like timeouts) and I think final replies to INVITE. Those will need
to be ironed out.
Please test it out and report any issues.
Regards,
Ovidiu Sas
On Fri, Dec 3, 2010 at 3:57 PM, Jeff Pyle wrote
.
Regards,
Ovidiu Sas
On Fri, Dec 10, 2010 at 12:20 PM, Duane Larson wrote:
> I am not sure if I did this right.
>
> At first I just replaced the subscriber.c file and tried to compile. That
> was my last email.
>
> Now I did the following
> svn co https://opensips.svn.so
Have you also updated the layout of the presentity table?
Check for the extra_hdrs field.
Regards,
Ovidiu Sas
On Fri, Dec 10, 2010 at 12:52 PM, Duane Larson wrote:
> I did update the presentity table to be 5 instead of 4. Perhaps i didn't
> clean up the previous install. Let
in main_loop (argc=, argv= optimized out>) at main.c:872
> #12 main (argc=, argv=) at
> main.c:1388
>
>
> libc.so.6 keeps popping up in the core dumps.
>
> On Fri, Dec 10, 2010 at 12:15 PM, Ovidiu Sas wrote:
>>
>> Have you also updated the layout of the pre
rd to fix this
issue).
Or maybe this is the proper fix (if the free is already done by the
underlaying mi implementation).
Regards,
Ovidiu Sas
On Fri, Dec 10, 2010 at 4:02 PM, Duane Larson wrote:
> Thanks for working with me on this Ovidiu. I think this is what you are
> wanting (sorry no
There are a few things that needs to be fixed before the release:
- final replies for INVITE does not show up in the b2b_entities reply route;
- sometimes there are retransmissions for PUBLISH requests (I will
provide you with logs for this one).
Regards,
Ovidiu Sas
On Tue, Dec 14, 2010 at 8
Here's how to investigate/debug memory issues:
http://www.opensips.org/Resources/DocsTsMem
Regards,
Ovidiu Sas
On Thu, Dec 16, 2010 at 12:44 PM, Ronald Cepres wrote:
> Hi to all,
> If I run OpenSIPS (1.6.3 ) for a long time while calls are coming in, it
> suddenly stops with
See the textops README file:
http://www.opensips.org/html/docs/modules/devel/textops.html#id250476
Regards,
Ovidiu Sas
On Mon, Dec 27, 2010 at 2:47 PM, David J. wrote:
> I saw a new feature for detecting on hold;
>
> Where could I see a small example of this?
&
Make sure that the "domain" feature is off (as it is by default):
http://www.opensips.org/html/docs/modules/devel/auth_db.html#id250089
Regards,
Ovidiu Sas
On Sat, Jan 8, 2011 at 3:24 PM, Chris Liu wrote:
> Hi All,
>
> I am new to opensips. I'd like to let users be abl
Just enable the nathelper module along with rtpproxy configured in
bridge mode and it will work fine.
Regards,
Ovidiu Sas
On Thu, Jan 20, 2011 at 10:45 AM, Alessandro Illiano
wrote:
> Hi Bogdan,
> Unfortunately i don't know... the problem is that we have many carriers
> connec
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
>
>
>
>
hange visible to the b2b module is the RURI.
Regards,
Ovidiu Sas
On Wed, Feb 2, 2011 at 12:52 PM, Kamen Petrov wrote:
> 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
>
ips B connects to the termination
> 4) the RTP goes between the softphone -> opensips A -> rtpproxy
>
> Would that work ? :)
>
>
>
>
>
> On 2 February 2011 20:04, Ovidiu Sas wrote:
>>
>> The nathelper module is performing changes on the received INVITE
>>
Hello Ryan,
Please try the latest version from trunk.
Please test and report back.
Regards,
Ovidiu Sas
On Thu, Feb 3, 2011 at 3:33 PM, thrillerbee wrote:
> Anca,
> Would it be possible to alter the built-in top hiding module so it doesn't
> strip the from display name?
&g
Maybe we should deprecate the flag, to avoid confusion in the future
and keep the code simpler.
Regards,
Ovidiu Sas
On Fri, Feb 4, 2011 at 12:32 AM, Bogdan-Andrei Iancu
wrote:
> Hi Ronald,
>
> there is no problem with using the flag and create_dilalog() in the same
> time - the di
There are several db supported by opensips.
For db_text and db_berkley, a path to the location of the db may be provided.
For mysql/postgres, the name of the db may be provided.
All this options can be specified in the opensipsctlrc file.
Regards,
Ovidiu Sas
On Fri, Feb 4, 2011 at 11:20 AM
Have you checked if the your config is altering the SDP?
Check the SDP for the same message before coming to your opensips
server and after leaving the opensips server.
If the SDP is altered and the IP in SDP is pointing to your opensips
server, then there is your problem.
Regards,
Ovidiu Sas
For now, best thing to do is to separate functionality:
- one server doing topology hiding;
- one server doing routing, accounting, rtp proxy, etc.
Regards,
Ovidiu Sas
On Sun, Feb 6, 2011 at 9:23 AM, Maciej Bylica wrote:
> Hi,
>
>> I am running Opensips 1.6.3 and trying to do top
You are re-posting the same question again without providing any
additional info:
http://lists.opensips.org/pipermail/users/2011-February/016626.html
Regards,
Ovidiu Sas
On Mon, Feb 7, 2011 at 12:20 PM, Chris Stone wrote:
> We have an Opensips 1.4 installation that routes calls to multi
d
and your RTP problem is somewhere else.
Regards,
Ovidiu Sas
On Mon, Feb 7, 2011 at 1:10 PM, Chris Stone wrote:
> On Mon, Feb 7, 2011 at 10:35 AM, Ovidiu Sas wrote:
>> You are re-posting the same question again without providing any
>> additional info:
>> http://lists.opensip
By default, opensips does not modify the SDP.
Double check your config. If you don't need to touch SDP, make sure
that you are not loading nathelper or mediaproxy. Those are the two
modules that are changing SDP.
Regards,
Ovidiu Sas
On Mon, Feb 7, 2011 at 5:39 PM, Chris Stone wrote:
>
On Mon, Feb 7, 2011 at 9:14 PM, Chris Stone wrote:
> Sorry all for the last message - too quick on the Send button.
>
> On Mon, Feb 7, 2011 at 6:48 PM, Henk Hesselink
> wrote:
>> Hi Chris,
>>
>> That config should't touch the Contact header, and yet that's also been
>> modified:
>>
>> In: C
ee it this is fixing your issue.
Regards,
Ovidiu Sas
On Tue, Feb 8, 2011 at 2:51 PM, Chris Stone wrote:
> Dave,
>
> On Tue, Feb 8, 2011 at 12:02 AM, Dave Singer
> wrote:
>> Don't know what tools you are familiar with so here are some
>> suggestions for what they&
mask everything on the SDP
side (see the 'o' flag for rtpproxy commands).
Regards,
Ovidiu Sas
On Tue, Feb 8, 2011 at 2:40 PM, Jeff Pyle wrote:
> Hello,
> Since the top-hiding scenario doesn't touch the SDP, it seems some
> extracurricular textops may be required to f
counter
and you have a second Via with exactly the same branch parameter.
Regards,
Ovidiu Sas
On Tue, Feb 8, 2011 at 3:58 PM, Chris Stone wrote:
> Ovidiu,
>
> On Tue, Feb 8, 2011 at 1:16 PM, Ovidiu Sas wrote:
>> Based on your description, it seems that you are dealing with a wei
e of beast then a proxy.
Every time a message is relayed in b2b mode, there are actually two messages:
- the received one:
- the new one that is created and sent.
Lump changes applied to the received messages are not visible to the
newly created message.
Rega
In b2b routes you see the received message and therefor you cannot
apply any changes to the message that is sent (you don't have a handle
to it).
In local route, you see the INVITE that is about to be sent out and
that one is modifiable.
Regards,
Ovidiu Sas
On Tue, Feb 8, 2011 at 4:48 PM,
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