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> On 11/14/2016 11:39 PM, Saúl Ibarra Corretgé wrote:
>> Hi there!
>>
>> It’s that time of the year again, time for FOSDEM paper submissions! Next
>> FOSDEM we’ll have a “Real Time Communications” devroom, which is a good fit
>> for OpenSIPS and
/pipermail/fosdem/2016-October/002481.html
I’m part of the organising team, so please do reach out if you have any
questions / problems.
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h MediaProxy? Also, how difficult is it to migrate
> to MediaProxy from within OpenSIPS?
>
MediaProxy is not designed to work behind NAT, except in 1-to-1 NAT scenarios
such as Amazon EC2. Check the advertised_ip setting in the configuration file
(config.init).
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log line that SSL2 is needed?! The 2.0 refers to the
SIP version, not the SSL/TLS version.
SSL2 was deemed as insecure very long ago and OpenSIPS no longer supports it.
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python version that Centos 6
> uses or from other packages (twisted?) may be the cause but we don't discard
> other options (accouting, some sysctl fine tunning option...)
>
> Any ideas would be appreciated.
>
> Thanks!
>
>
> Koldo Aingeru Marco
cket-2-1
>
> Best regards,
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> http://www.opensips-solutions.com
>
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"
> and "falid to get local sdp" in linphone.
>
This has nothing to do with OpenSIPS. Your clients need to have DTLS support if
one of them is a WebRTC endpoint, or you’ll have to use a gateway. OpenSIPS +
OverSIP will just get the signalling working.
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ell users they are doing it wrong, iMHO.
> Of course, we can detect that at script parsing and decide how to handle it:
>- ignore and do it blocking
>- warning and do it blocking
>- error and do not start.
>
I’d say we shouldn’t start :-)
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On 03 Sep 2014, at 11:59, Bogdan-Andrei Iancu wrote:
> On 03.09.2014 12:06, Saúl Ibarra Corretgé wrote:
>> On 02 Sep 2014, at 18:26, Răzvan Crainea wrote:
>>
>>> Hi all,
>>>
>>> Among the last discussion of the last IRC meeting[1] was related t
e Facebook page seems to be a person called “Opensips Server", not an actual
page people can “like”. That was common practice before Facebook introduced
“Pages”, but not anymore.
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sync OP could not be done.
I find this quite problematic. How does John Doe know a function supports async
or not? Maybe the configuration checker can know this and not let OpenSIPS
start if you are doing it wrong? Alternatively, blocking functions could be ran
in a thread pool thus
l?
> Since we don't have a final decision for this topic, we are looking for more
> input from you guys.Anybody is welcome to throw any kind of useful feedback
> on this matter, so don't be shy!
>
IIRC this is not standard, and Apple uses it somewhere on their FaceTime
t; are good? How do I debug RTP?
>
You should see syslog entries about MediaProxy being started. What do you mean
by “debug RTP”?
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is is a great initiative folks, kudos! I’ll try to make it to the meeting.
Can I get “Asynchronous TLS” added to the list? :-)
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me)
> File "/usr/lib/python2.7/ctypes/__init__.py", line 383, in __getitem__
> func = self._FuncPtr((name_or_ordinal, self))
> AttributeError: /usr/lib/x86_64-linux-gnu/libgnutls.so.28: undefined symbol:
> gnutls_certificate_verify_peers3
>
hat?
>
This is not possible with MediaProxy. You could alter the routing in your
machine to use a different gateway when traffic is directed at host X, but it’s
out of the scope on MediaProxy.
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The request is not sent until later, when the request will actually go out.
Changing it to behave like you want would imply “reserving” the relay early,
and using that info later, it’s not a trivial change, and unfortunately I can’t
allocate time to look deeper into this at the moment, sorry.
ed_sdp.
>
You could use the individual functions: ude_mediaproxy and end_media_session,
but then you need to cover all scenarios that engage_media_proxy already does
automatically.
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HI,
engage_media_proxy just sets come callbacks in the dialog module, so the actual
SDP changes happen asynchronously, later.
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ke caller_codec, callee_codec,
> caller_remote etc. Is this possible and if so how can I add this fields
> (like extra settings in config.ini?).
>
IIRC you’d have to modify the source code for that. Or use RADIUS, which uses
separated fields.
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signature.
to run MediaProxy within an
> EC2 like environment (ie, public/private ip)?
>
Check the config.ini.sample file, it’s the advertised_ip setting.
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ess of our MediaProxy server.
>
> Looking at the OpenSIPS documentation for MediaProxy module I saw
> nothing of the sort.
>
The engage_mediaproxy function (or use_media_proxy if you use the separated
ones) will do it automatically.
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Opensips, Mediaproxy, all deps,
> ...), or find a definitive misbehaving client, I'll restart the relays on a
> schedule. If I am able to reproduce the exceptions I'll provide debug data.
>
Ok, thanks! If you manage to get some SIP captures feel free to email them to
me private
daemon, which, of course, needs to be running. Do you
see any errors on syslog when you start mediaproxy-dispatcher? You can also try
to start it in foreground, which will spit the output to standard output:
mediaproxy-dispatcher —no-fork
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ne, since it’s
not used for routing.
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this is a known bug that hasn't
> bitten us until today? Or, something else?
>
> They are scheduled for replacement with more current software, but until
> then, I'd like to learn what I can.
>
I browsed through the MediaProxy changelogs just in case memory
generate your own certificates with
> http://tinyca.sm-zone.net/, but I think this project is gone…
>
On Debian it’s packaged as “tinyca2”, maybe your distro provides a similar
package?
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> ERROR:tm:msg_send: tcp_send failed
>
> Any guess, what could be the reason?
>
OpenSIPS is trying to establish an outgoing TCP connection, and it’s not able
to do so.
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t;
> / Ionut Muntean
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iacontrol.py, in the update_session and/or update_media functions.
Let me know how that goes!
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the right direction :)
>
What "carrier grade" features are those?
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On May 2, 2014, at 2:50 AM, Nick Altmann wrote:
> It seems no problem already. Everything works as expected.
>
Thanks for confirming Nick! I'll cut a bug fix release soon.
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close your call. The solution is not
>> using engage_media_proxy, instead using use_media_proxy/end_media_session
>>
I managed to catch (and hopefully fix!) the bug. We tested it and it works, but
some extra testing would be nice. You can find the patch attached, which can
lly for you. However, you can set the
following AVP [0] to manually choose the relay. You can write your logic in
OpenSIPS and set it before calling engage_mediaproxy.
[0]: http://www.opensips.org/html/docs/modules/devel/mediaproxy.html#id249322
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ImportError: No
> module named util
>
>
> I have python 2.6.6 installed , if any one has any suggestions they will be
> great.
>
You'll need an old python-application as well, since the API has changed. Check
the version it depended on (in the debian/control f
> Apr 1 16:29:09 localhost media-relay[14388]: (Port 35004 Closed)
> Apr 1 16:29:09 localhost media-relay[14388]: (Port 35005 Closed)
>
I see the streams are marked with "ICE". I'll have a look at the logic.
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diaproxy thinks it's not direct connection and breaks it, when clients
> set up direct connection before?
>
> What do I do wrong?
>
You don't seem to be doing anything wrong. Can you paste the debug logs of the
relay?
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t are not
> working and those are not based on the SIP SIMPLE SDK.
>
> Is it possible to tweak the structure of the element in OpenSIPS
> 1.10?
>
It's not possible. You'll have to modify the C source. If you make a patch
which moves the namespace declarat
attributes are not defined?
> Also the encoding Header is missing. This is a bad style.
>
I'm not sure when the encoding went, but yes, it should have been there, that's
a bug.
> I don't find any UA that works with this
advertised_ip =
Just put your public IP address there and the OpenSIPS module will put that one
on the SDP.
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t, or direct
> us to migration installation config, parameter setting documentation
> kindly. Pretty please :)
>
Looks like what you want is to advertise the WAN address some times and the LAN
address some other times. That's currently not possible.
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s.
TLDR: yes, that should work.
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address in the SDP, even
if MediaProxy is behind the NAT.
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> Best regards,
> Samuel.
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ppens if you run the binaries
(media-dispatcher and media-relay) manually?
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/ disable media relying. It can work with parallel forking, as
long as you don't have multiple early media replies. In that case IIRC media
proxy would pick the last one.
You can call use_media_proxy from the branch route in case you want to
implement your own logic.
Regards,
wrong?
>
You have done nothing wrong :-) We don't provide packages for Ubuntu Precise.
We will build them with the next release, but we have no ETA for that. You can
build it yourself by grabbing the code and doing:
debuild -us -uc
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e-chat-18), you can use SylkServer,
an Open Source implementation of it (among other things). http://sylkserver.com/
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ind anybody experiencing
> the same problem)
>
> used apt-get sources:
>
> debhttp://ag-projects.com/debian stable main
> deb-src http://ag-projects.com/debian stable main
>
Those repositories are for Debian stable. You are using Ubuntu, so you should
use the appropriate
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nce again, your feedback on these matters is important to us.
>
>
> Best regards,
>
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> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
>
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that's how MediaProxy communicates with OpenSIPS for ending dialogs. Extending
it to support other fifo methods should be pretty straightforward.
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the
> lines involving the presence_xml module and everything worked fine, but sadly
> I DO need this module.
>
> What is that due to? It's there a bug?
>
> regards,
> Estefania
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diaProxy source, it contains a simple website
which senders active calls.
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ith a step
by step tutorial. Also, "did not work" is not enough for anyone to help you. If
you want good answers you'll have to ask better questions.
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Hi,
Yes, examples are indeed wrong. I created an issue on GH which I'l address soon.
Regards,
On Sep 12, 2013, at 7:07 PM, Bakko wrote:
> hello,
>
> this means is there a error on residential script?
>
> Regards
>
> El 12/09/2013 11:44, Saúl Ibarra Corretgé escrib
gt;
> Any hint?
>
> Thank you
>
> Regards
>
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odules for OpenSIPS, to generate RFC 3680
payloads.
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ce_xml.so
>
If you want to handle the 'presence' event you'll need the presence_xml module.
You'll also need the xcap module unless you set force_active to 1.
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t;
> modparam("presence", "db_url","mysql://ossips:xxx@localhost/opensips")
> modparam("presence", "server_address", "sip:172.x.x.x:5060")
>
These look fine, did you load the db_mysql beforehand?
>
>
>
>
> 201
CRITICAL:core:yyerror: parse error in config
> file /etc/opensips/opensips.cfg, line 87, column 16-17: Invalid arguments
> Aug 28 08:39:17 sipia opensips: CRITICAL:core:yyerror: parse error in config
> file /etc/opensips/opensips.cfg, line 88, colum
t!
Also, for those with nostalgia, this actually works:
svn checkout https://github.com/OpenSIPS/opensips
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"exclude_modules?=" line
> - just changing "Configure Install Prefix" to "/usr/local"
> - "Save Changes"
> - gmake install
>
> The result is the same:
>
Doh, I had forgotten to backport that fix, I just did. Can you pull and try
again?
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help
> Torsten
>
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B for authentication data, for example by pointing
it to your OpenSIPS subscribers table.
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doing what we are
> doing then. I just hoped to not have to run a proxy, and to have native
> support in OpenSIPS for Websockets, that would be great :)
>
Indeed, but we need better TCP support beforehand, which will be part of
OpenSIPS 1.10. Also, patch
or
> functions and parameters.
>
> Am I missing something here? Any help pointing me in the right direction is
> much appreciated.
>
OpenSIPS does not support WebSockets as a transport for SIP. It does support
acting as a core proxy behind a WebSocket capable edge proxy, which
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
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h SourceForge and GitHub). However, we believe that
> this is an acceptable compromise.
>
I think it's a good thing to do, this will probably also help in removing old
tickets that have been lingering around for too long ;-)
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does.
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use the user-input element from RPID to set the last
activity timestamp, which could act as a tie breaker, for example.
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problem from opensips side?
>
What if I publish different presence states from different devices for the same
user? OpenSIPS will aggregate the bodies in a single NOTIFY just fine.
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state still in the table,Is this correct?
>
The state will remain in the table as long as it's not expired or someone sends
a PUBLISH with expires 0. If the application exited abnormally the information
will remain in the table until it expires.
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fine
> PERM_MAX_SUBNETS 128 in hash.h)
>
> My questions are:
>
> ¿Is this networks limit real? ¿Can I increase it? ¿how?
>
I feel your pain, I hate compile time limitations with a passion. Right now the
only way around is to modify that define
> MSN: shari_78...@hotmail.com
> Email: shaherya...@googlemail.com
>
>
>
> --
> Mit freundlichen Grüßen
> Muhammad Shahzad
> -------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +49 176 99 83 10 85
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On May 14, 2013, at 10:10 AM, arra wrote:
> a libgcrypt11 package where I can download it，I think I need a web link to
> download it ,thank you!
>
You can get the 'ubuntu0.1' packages for your architecture from here:
http://ag-projects.com/ubuntu/pool/main/libg/li
he news seem to be repeated: they are displayed as
headlines at the top of the page and also on the bar to the right. I'd leave
only the bar on the right, so the first thing newcomers see on the website is
"What OpenSIPS is".
My 2 cents :-)
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ng how to iterate through a list of headers with the
same name.
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e ?
>
You can use the following transformation:
var counter = $(hdr(Diversion){param.value,counter}{s.int});
That will give you the integer value of the counter.
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Use
b+17000]
> Apr 26 11:33:00 ubuntu kernel: [63156.141836] media-dispatche[23060]:
> segfault at c ip b7700cb7 sp bfeb00a0 error 4 in
> libpthread-2.15.so[b76f8000+17000]
>
We provide a libgcrypt11 package in our ubuntu repository, install it from
there and you shouldn't run in
currently still provide Debian packages for Debian stable
(squeeze) and unstable. Both 32 and 64 bits.
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..@googlemail.com
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> fixed the docs page.
>
Sorry about that, I missed to link the modules documentation. Thanks for fixing
it, Vlad.
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he updated state. Updating
resource-lists also triggers a call to refreshWatchers, however, because
resource-lists doesn't change the authorization state you'll not see a NOTIFY.
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t the document has changed, you
can subscribe to the xcapdiff event.
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a fork and
anyone can make inline comments on the diff. Then the contributor can overwrite
those commits (in his fork) until everything looks nice. Then, a committer can
merge the patch in, but the authorship is preserved so the patch author gets a
nice attribution in the form of bei
a you need your endpoints to use ICE. MediaProxy is ICE aware,
so if it detects that media is flowing P2P it will bail out. You can find more
information about this here:
http://mediaproxy.ag-projects.com/projects/mediaproxy/wiki/ICE
Regards,
--
; document,
> and then Bob received the presence.winfo Notify message.
> So which condition trigger the Notify message of presence.winfo in opensips?
>
presence.winfo notifications are triggered when an active subscription changes
authorization state. By active I mean a SUBSCRIBE was sent and a 2XX response
was received. Since you modified resource-lists and, unless you are using the
OMA mode, it has nothing to do with authorization, there is no presence.winfo
event to be generated.
Regards,
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Saúl Ibarra Corretgé
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Time to make some new builds :-)
Cheers,
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Saúl Ibarra Corretgé
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lid is ok
> and the error no longer appear in log. So all work fine :)
>
I'm glad it works for you!
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Saúl Ibarra Corretgé
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regards,
PS: If you are thinking about doing a lookup() for the location, that won't
work in case the user re-registers while in the middle of the dialog, if Path
is used and a different path was chosen. I can elaborate on that but I didn't
want to complicate the e
txt file.
>
>
I'm not sure if there is any problem in using ropology_hiding with callcontrol,
if there is I'm not aware of it, but other than that, I don't see any problem
with those steps.
Please do check that misrouted INVITE, I think it's the actual cause for this
quest (481) the proxy is confused
and so is CallControl.
Can you better describe how you are doing the routing and when do you call
call_control in your OpenSIPS script? (an overview, please don't attach a full
configuration file).
Regards,
--
Saúl Ibarra Corretgé
A
ouldn't be harmful.
Recap: please apply the supplied patch and run these tests:
- Just run the current config, watch for the log lines and see if audio flows
regardless
- Set userspace_transmit_every = 4 in the [relay] section of the configuration
file
Please let me know how testing goes.
Regards,
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Saúl Ibarra Corretgé
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Allow: NOTIFY, CANCEL", but still
> get INVITE msgs.
>
Checkout the 'm' flag in the lookup function:
http://www.opensips.org/html/docs/modules/devel/registrar.html#id293002
Regards,
--
Saúl Ibarra Corretgé
AG Projects
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User
can't see the relative
> session on Cdrtool table.
>
> Do you need only call-controll log or opensips logs too? If is needed i can
> add some extra log in my opensips.cfg for better debugging.
>
The CallControl logs and a SIP trace for the call should be enough.
Re
call ends? Please post a bigger chunk of the
syslog output so that we can get the context of the problem.
Regards,
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Saúl Ibarra Corretgé
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On Feb 20, 2013, at 9:41 AM, Jacek Konieczny wrote:
> On Tue, 19 Feb 2013 15:45:01 +0100
> Saúl Ibarra Corretgé wrote:
>>>- file download system
>>
>> Sort of. You can't upload files, but it will generate tar files for
>> each tag.
>
>
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