versions of mediaproxy, when trying to
build 2.2.3, read the INSTALL, and installed the needed dependencies,
configure script told that all was ok, but mediaproxy does not compile giving
me the errors I paste on the previous mail.
Any other clue?
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+Asterisk+A2Billing and IHMO that's a
horrible solution, with lot of problems on you future.
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it from failure_route.
For example, instead of something like:
failure_route[1]
{
if()
{
setflag(15);
}
Just do:
route(XX):
{
setflag(15);
}
...
failure_route[1]:
{
...
if(...)
{
route(XX);
...
}
...
}
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... but AFAIK, accounting is taking into account when the
positive reply comes, that is .. on an INVITE, the acc record is saved when
the 200 OK comes back from the UAS.
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On Tuesday 02 June 2009 01:11:02 Iñaki Baz Castillo wrote:
El Martes, 2 de Junio de 2009, Raúl Alexis Betancor Santana escribió:
On Monday 01 June 2009 22:37:58 Charles Solar wrote:
I have tried sending messages by append_branch and send, and from
packet captures and xlog I know the flags
On Martes, 2 de Junio de 2009 10:08:33 Iñaki Baz Castillo escribió:
El día 2 de junio de 2009 8:30, Raúl Alexis Betancor Santana
r...@dimension-virtual.com escribió:
Not exactly. You choose when and where to do the accounting. In fact,
logging failed calls (busy, cancelled since no answer
On Tuesday 02 June 2009 19:51:09 Iñaki Baz Castillo wrote:
El Martes, 2 de Junio de 2009, Raúl Alexis Betancor Santana escribió:
That's imposible to do, not only because it will be a very poorly
accurate accounting but also because the ACC module does not account
based on the initial INVITE
.
We rung mediaproxy 2.3.4 without any problem in our systems.
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On Lunes, 8 de Junio de 2009 12:30:30 Gavin Henry escribió:
2009/6/8 Raúl Alexis Betancor Santana r...@dimension-virtual.com:
On Lunes, 8 de Junio de 2009 10:01:58 Gavin Henry escribió:
We have the same problem on another server and can't resolve it.
Just and advise:
run it like
opensips[22591]:
Only this .. is telling you that something is not ok with the network config,
maybe you have firewall rules activated, or you are trying to rewrite some
address and using one that is not set up on the interfaces.
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will have no luck trying to run mediaproxy on that host.
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the request, but do I need to cleanup
the dialog somehow?
Thanks,
Brett
You could not do that without a B2BUA, because you will break CSeq on the
stablished dialogs
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On Thursday 11 June 2009 22:57:14 Brett Nemeroff wrote:
2009/6/11 Raúl Alexis Betancor Santana r...@dimension-virtual.com
On Thursday 11 June 2009 22:27:23 Brett Nemeroff wrote:
Does anyone know if there is an actual QSIG implementation of REFER?
That
may make this whole issue moot
NIC
card?
Anyone has experience about this problem pls jumbing,
I'm sure at 99,9% that the error is caused by some firewall rules you have on
that machine, or a network missconfiguration.
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If anyone is interested on this .. this little, little patch.
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diff -rN old-callcontrol/call-control new-callcontrol/call-control
55a56,58
parser.add_option(--config, dest=configuration_filename,
default
, and using
the same line for Inet navitation + VoIP, over a 3M/320Kb line you could
drive 5 G729 calls without any problem, and over a 10Mb/1Mb line, more than
12 ... so I don't undestand where you have problems with a 1.5Mb/1.5Mb line.
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this system?,
Can I use SEMS?
For building a voipmail system, better to use SEMS than Asterisk, less
resources needed, more concurrent calls.
Anyhow, all depends on your budget and how do you need it to scale.
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Sorry I must answer on this way .. but as the list don't set the reply-to
correctly and this usser's mail server refuses direct mails ..
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---BeginMessage---
On Tuesday 07 July 2009 20:15:01 li...@grounded.net wrote:
On Tue, 7 Jul 2009 14:02:11
into lot
of troubles trying to do something with a SIP PROXY
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doubds, post confings and traces of what you don't get running
and I'm pretty sure you will get answers that will help you, but if you
insists on doing too general questions you will allways get the same
answers.
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) ?
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with a Sipura or Polycom registered directly to
Opensips. It's just much harder to test because I don't have any extra
public IPs available in my home lab.
For properly handling the re-invite, did you call force_rtp_proxy INSIDE the
in-dialog procces ?
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On Sunday 12 July 2009 00:48:26 Jeff Pyle wrote:
On 7/11/09 5:33 PM, Raúl Alexis Betancor Santana
r...@dimension-virtual.com wrote:
I will not question why are you trying to use Mediaproxy if not for NAT
fixing .. X-)
Because I can. :) Or, rather, I thought I could... Real world
, and if I need some blending
edge soft I backport it to stable.
Backporting mediaproxy it's a very simple task, as it's a very simple
dependency software.
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-application, as far as I remember)
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installing on the production ones.
That's all.
Most of times it's a matter of getting the .tar.gz, .diff and .dsc files of
the packages we need to port and then on the build system:
dpkg -x package1.dsc
cd package1-1.0.0
dpkg-buildpackage
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the
system (on a normal way, that does not occurs, but you don't have any warrant
of so)
But this is only a matter of POV's.
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system to Debian unstable
and leave other applications running with critical bugs.
Most, if not all AG app's could be used on Lenny or Sid with only updating
python-application.
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php.ini files
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don't use OpenSIPS) and B2BUA features on Callweaver
GW's
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Please ... could be possible to setup the mailling list as normal mailling
lists that with Reply just send the reply to the list and not to personal
inbox .. ?
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---BeginMessage---
On Tuesday 04 August 2009 21:08:03 you wrote:
Maybe it's
it will allways receive
a copy from the list) ... I could allways use Reply All.
Well .. that's just my opinion.
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On Wednesday 05 August 2009 00:07:30 Dan Pascu wrote:
On Tuesday 04 August 2009, Raúl Alexis Betancor Santana wrote:
I wonder how other could use dialog module and other modules that depends
on it with such limitations ...
That's because you lack imagination and because your purpose doesn't
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to say that
could help (I said that dialog module have limitations, I said that I tested
it, so next step is to try to fix that limitations and send a patch, that's
something I have on my TODO)
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it. It's bad for business, it's bad for both products,
it's bad for everyone. NOBODY wins if commercial adopters see this kind
of petty bickering and egotism, especially from lead developers and
other significant stakeholders in the commercial ecosystem.
Full Agree.
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hard to reply to the poster. Since
neither reply to reply-all will get you the email of the poster.
I don't know any mail client that when you hit reply all, don't send the
reply to ALL, including list and poster, at least all common MUA does it that
way.
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that why when you do a Reply all with gmail, it
doesn't include all address on the reply.
All this things about gmail are well documented on Google LABS forums, they
have been requested as basic features by lot of users.
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On Wednesday 05 August 2009 17:09:56 Dan Pascu wrote:
On Wednesday 05 August 2009, Raúl Alexis Betancor Santana wrote:
On Wednesday 05 August 2009 14:58:32 Brett Nemeroff wrote:
And if you set to The List, it's very hard to reply to the poster.
Since neither reply to reply-all will get
mediaproxy-relay to be able to call it with a
diferent .cfg as the default one, have diferent listen ports and no more.
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think that a simple param like -c configfile is
better.
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On Thursday 10 September 2009 22:25:39 Dan Pascu wrote:
On 10 Sep 2009, at 18:00, Raúl Alexis Betancor Santana wrote:
So ... killing flies with bazookas ...
So ... when having to update, need to update n instances, instead of
only
one ..
So ... wasting hard disk (I know it's cheap
the way I want.
It works, maybe you have not set some options of the modules that let it works
but It works, trust me ;-)
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well ... our you NAT router
it's doing some kind of SIP ALG and disturbing the normal working of the
proxy.
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the polycom on the 401
response ... that's why you polycom do not send it credentials and never get
registered.
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UAC will refuse that
information and you will be wasting your time and resources.
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-Client ... better to follow the standars and just
use SUBSCRIBE to retreive other users status
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On Jueves, 15 de Octubre de 2009 00:31:10 Daniel Goepp escribió:
Route: sip:75.101.136.125;lr
Have you declared as a local domain?, if not ... OpenSIP will try to
route it, so thats were you have the loop of the ACK.
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are wright, that will be non-RFC conform ... moreover .. I don't undestand
why your subscriber needs that ... because Asterisk is non-RFC conform on lot
of things .. but not on that one.
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Ross Beer escribió:
Hi,
I have registrations replicated over two servers using a MySQL cluster
and have also tried replication at SIP level however I can not call
between servers.
The setup is as follows:
Soft phone A Reg OpenSips A
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means that you are looping yourself. Have you checked
with ngrep that you are not routing the petition over yourself again and
again ?
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Peter P GMX escribió:
Hello,
I tried the 302 redirect now in 2 different methods:
$ru = sip:06912345...@yy.yyy.yyy.163:5062; # rewite complete
contact address
rewritehostport(yy.yyy.yyy.163:5062); # rewite host
and port
and then
sl_send_reply(302, redirected);
In
Bogdan-Andrei Iancu escribió:
Peter,
You can add a header with whatever name, depends more on FS what kind of
header is able to use to take the src IP from.
Regards,
Bogdan
That will not run, FS as Asterisk, Yate and most of B2BUA/PBX outthere
..takes socket IP for validating it's
Bogdan-Andrei Iancu escribió:
So, in such a case, a real working solution will be to have opensips to
spoof the source IP of the outbound request? but this may really break
the transaction Only if the spoofing is done with skipping the VIA
insertion.
Regards,
Bogdan
Spoofing
easier, faster, cleaner and scales better, if you use
allow_trusted() from the opensips.cfg and work with your whitelist from the
proxy POV, it's just and advice ... you will sleep better ;-)
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a very basic
problem, that many people probably have faced.
Could you please recommend me a way to troubleshoot this issue?
Thanks a lot,
Justin.
Some SIP trace would be nice to begin ...
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any problem.
Are you using the dialog module? ... if yes, take into account it limitations
to work with parallel forking.
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/Snom phones connect without issue.
If anyone could offer some advice it would be much appreciated as I'm
tearing my hear out :-)
Better if you attach some SIP trace of a working an a non working call
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NEEDS the dialog module ... and it's a
know limitation of dialog module that it doesn't work AT ALL with parallel
forking, or with multiple 1XX replies.
Better if you limit your uses of the dialog module to the minimum ... let
say ... to 0 .. ;-)
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.
No, it's not a bug in mediaproxy. I think it's a bug in engage_media_proxy(),
I'm pretty sure ... because I use mediaproxy with parallel forking without
any problem on the same scenario, but using use_mediaproxy
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to me that i forgot something in my config, a refer module
or something? any toughts/pushes in the right direction would be greatly
appreciated!
As far as I see ... 77.73.226.254 it's refusing to do the REFER, what are
you inside IP's, and what are you outside one ? ...
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Peter den Hartog escribió:
77 is the sip trunk, 90 are my IP's.
So, it's your trunk provider the one who refuses to do the REFER
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use_media_proxy() then? On the 1st INVITE
or later with the 200 OK with SDP? Is there any example out there?
On First invite after checking if needed, on_reply for 1XX or 200 with SDP, on
re-invites
You could see and example on sipwise.com
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?, that could explain the problem.
This INVITE is the one that just arrive at your proxy? ... if so .. your NAT
router is doing sip-alg .. so mangling all the SIP dialog, and they usually
does a very bad job with that.
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Uwe Kastens escribió:
Raúl Alexis Betancor Santana schrieb:
On Friday 23 October 2009 14:14:52 Uwe Kastens wrote:
Hi,
Is there an option to prevent this behavior with mediaproxy?
opensips: 10.20.20.159 and 10.20.30.159
UACs: 10.20.20.25 and 10.20.20.26
UAS: 17.17.17.167
Uwe Kastens escribió:
I was not able to get it up and running with use_media_proxy and
end_media_session, since the mediasession was ended, if a BYE for the
2nd branch arrived.
That could only occurs if you get something wrong with the branches. For
sure :-)
I changed a lot in my script
Uwe Kastens escribió:
Yes, I am sure about this. Is there any good tutorial out there?
No, as far as i known, only the docs about how branches should work.
no, sorr, a typo, should be:
new:
route[1] {
...
engage_media_proxy();
...
t_on_branch(1);
t_on_reply(1);
route(3),
...
}
for
OpenSIPs who can speak to that rtp proxy.
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On Tuesday 03 November 2009 12:19:36 Olle E. Johansson wrote:
3 nov 2009 kl. 12.36 skrev Raúl Alexis Betancor Santana:
On Tuesday 03 November 2009 10:15:31 michel freiha wrote:
Dear All,
I would like to ask you please if there is a way to use OpenSIPs
with a
proxy for rtp other than
]
- INVITE
- INVITE
180 Ringing -
Maybe you are rerouting the second invite back to your UAC, or you do some
aliasing ... or something similar.
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number. For example IP phones that are using STUN.
Or a router doing SIP-ALG and not working ...
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no improvements
on running more that one instance per server.
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On Saturday 28 November 2009 17:50:32 Brad Bendy wrote:
If it's all done on the kernel level, then RAM won't really matter either
correct? No need for massive amounts, just good NIC cards and a decent
processor id assume, correct?
Yes, you are ok.
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no will be dropped, better .. it had been dropped ... ;-)
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not
planned to develop, I can try to implement this feature and provide
patch or modified source files. Will it be useful for someone else but
me? :)
I think that everyone off us use the group module for that task of ACL's, also
you could use a view as the table for auth.
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have just ordered my copy today ... ebook + paperback ... Just a very, very
good book Flavio, congratulations
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call)
For doing a MoH server, you will need to use the B2BUA module (and I don't
know if could be used for this ..).
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situations?
Dear VoipExpert ... let's talk first about how much you would have to pay for
that information, taking into account that it's obvious you don't want to do
the tests yourself, so this would be a payed assistance.
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that the one that wrote that artcile doesn't have any idea
about what he was writing, moreover taking into account that it doesn't give
any test results or either describe a test scenario, it seams only to talk
that way because someone else told him to do so ...
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, but that
patch doesn't add transcoding capabilities to RTPProxy, only to use
transcoding hardware.
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into account Asterisk limitations on
handling REGISTERS.
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On Monday 07 June 2010 14:20:56 mehdi boudou wrote:
Hello,
Is it possible to do multi and sequential ringing just with the
configuration file without any other module ?
Yes, lookup at the docs how to do serial forking and parallel forking
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.
pl let me know if I am not clear
If you paste an ngrep trace of what you get and try to explain more clear what
you whant, maybe we could help you.
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.
You was asking for a custom-ready-to-use solution, you was asking for free-
consultance, and thats what you received ... nothing.
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Does anyone tryed to use CDRTool + mediaproxy without Freeradius support? ...
I mean only using MySQL.
It's possible?
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regards
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On Domingo 05 Septiembre 2010 12:39:59 Saúl Ibarra Corretgé escribió:
On 09/05/2010 01:08 PM, Raúl Alexis Betancor Santana wrote:
Hi all,
I'm testing CDRTool 8.0.4 and getting some problems with it ..
Let first say what I have done:
- Install cdrtool 8.0.4 from darcs repo
or witchever traffic capture tool you
feal confortable with
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file, it does nat-fixing-handling, and when you undestand what
it does, try to modify it for adding what youe need.
Also a bunch of SIP knowleadge is a must.
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:
http://www.voip-info.org/wiki/view/Routers+SIP+ALG
And don't try to fix the request, because the problem is not only related to
Content-Leght, SIP-ALG implementations are full-broken
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On Martes 19 Octubre 2010 20:08:11 Dmitry Kravchenko escribió:
I have D-Link DIR-615. It's not in the list but SIP ALG is on by
default. My friend has the Linksys, not in the list.
Off course, not all models that have SIP ALG are on the list, it's a wiki, you
could add them.
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Raúl Alexis
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