Hi Borgan,
Sorry by trying to debug the problem I understood the hole picture. I
think it might be a bug or a feature request for the tm module.
The setup is:
PSTN-GW - opensips as statefull proxy - AST1 + AST2
If I make a call from pstn over the opensips to a specific SIP-URI, the
call will
Hi,
Are you controlling all servers? If so, you can implement a trust based
on IP Adresses in your script.
BR
Uwe
Pacho Baratta [fabbricadigitale] schrieb:
Hi all,
i’d like to know how should I do to place a call to an Opensips
requesting authentication.
This is the environment:
Unfortunately not, i have no authority on Opensips2.
fabbricadigitale srl
Pacho Baratta | Senior Systems Engineer
Tecnhology Engineering
-
Via A.Volta, 3 - 26041 – Casalmaggiore - CR
Phone +39 0375 284600
Fax +39 02 57760002
mailto:p.bara...@fabbricadigitale.it
Ok,
What kind of credentials do you have for opensips2? Individual for each
account at PBX1 or one account for all?
BR
Uwe
Pacho Baratta [fabbricadigitale] schrieb:
Unfortunately not, i have no authority on Opensips2.
fabbricadigitale srl
Pacho Baratta | Senior Systems Engineer
Hi Bogdan,
do you have any idea what Brad is doing different than I do? As I'm also
intensively using both functions (alone and together, depends on call
direction), I'm a little bit concerned. However, I did not (yet) meet
those crashes...
Cheers,
Thomas
NB: I'm using those functions only in
One for all should be ok.
Thanks, pacho
fabbricadigitale srl
Pacho Baratta | Senior Systems Engineer
Tecnhology Engineering
-
Via A.Volta, 3 - 26041 – Casalmaggiore - CR
Phone +39 0375 284600
Fax +39 02 57760002
mailto:p.bara...@fabbricadigitale.it
El Jueves, 22 de Octubre de 2009, Uwe Kastens escribió:
Hi Borgan,
Sorry by trying to debug the problem I understood the hole picture. I
think it might be a bug or a feature request for the tm module.
The setup is:
PSTN-GW - opensips as statefull proxy - AST1 + AST2
If I make a call
Hi,
which has ACKed one call already and will get a 2nd 200 OK with the
same branch but different call-id.
Different call-id? Perhaps yo mean different To tag as the Call-ID is
generated by the UAC (the gw) and must be the same in both legs.
You are correct. The Call-ID AND branch is the
El Jueves, 22 de Octubre de 2009, Uwe Kastens escribió:
What is exactly the issue? is the above explained by me?
Yes. I was able to step on testing and found out, that our reference
system (softsiwtch) is handling it correctly. The asterisk servers we
are using as mediagateways are unable
Hi,
Iñaki Baz Castillo schrieb:
El Jueves, 22 de Octubre de 2009, Uwe Kastens escribió:
What is exactly the issue? is the above explained by me?
Yes. I was able to step on testing and found out, that our reference
system (softsiwtch) is handling it correctly. The asterisk servers we
are
Hi,
system (softsiwtch) is handling it correctly. The asterisk servers we
are using as mediagateways are unable to handle it correctly - so I will
need a fix for them.
Anybody know if this has been fixed on asterisk?
I don't understand, why is noa an issue of Asterisk?
Isn't a problem in
El Jueves, 22 de Octubre de 2009, Uwe Kastens escribió:
I don 't understand... first you said that the GW send a call to OpenSIPS
and OpenSIPS to both AST1 and AST2 (Asterisk). And the problem is in the
GW because ignores the second 200.
Am I wrong?
The setup is:
asterisk(gw) opensips
0016e640d47e2c8bcd047677c...@google.com
Content-Type: text/plain; charset=iso-8859-1
Content-Transfer-Encoding: quoted-printable
MIME-Version: 1.0
Hi Duane=2C
=20
Here is my config and SIP trace.
=20
There is one interesting thing in the SIP trace=2C that is the SDP codecs. =
I have G711u=2C
Is there any way to do this?
fabbricadigitale srl
Pacho Baratta | Senior Systems Engineer
Tecnhology Engineering
-
Via A.Volta, 3 - 26041 – Casalmaggiore - CR
Phone +39 0375 284600
Fax +39 02 57760002
mailto:p.bara...@fabbricadigitale.it
www.fabbricadigitale.it
Hi,
Patched is installed now, going to push some load and see if I can crash it.
Thanks and ill keep you updated.
Bogdan-Andrei Iancu wrote:
Brad,
Please apply the attached patch and see if it's fixing the problem for
you.
Regards,
Bogdan
Brad Bendy wrote:
#0 0x000801517b39 in
Hi Every one,
I'm new to the opensips. I dont know wheter it is a right question to post
here. I worked on the amost all modules of opensips. I had some queries
regarding dialplan module.
1)What is the exact usage of dialplan module other than calling
regexpression from database.
I tried some
Hi,
After reading lots of docs and mailing lists, it looks like there is now
solution for asterisk available and looks like that might be a long way
till then.
Maybe its possible to implement that feature in TM?
http://www.codename-pineapple.org/doc/html/sip3_dialog_match.html
BR
Uwe
Hello,
Is it possible to use a view/table from a remote database (not from
opensips db) for authenticating users?
I'm testing following scenario but cannot get it to work.
I have a separate user provisioning system which store username, password
and domain in a external mysql db. I want
Hi Hasitha,
You need to have a table version, with the name of the table and the
right versions. Check in the source code the MySQL scripts who create
the databases. For each OpenSIPS table there is a record in the
version table.
Regards,
Flavio E. Goncalves
At 11:30 AM 10/22/2009, you
Thanks Flavio.
All works perfectly now. :)
On Thu, Oct 22, 2009 at 2:35 PM, Flavio E. Goncalves
fla...@asteriskguide.com wrote:
Hi Hasitha,
You need to have a table version, with the name of the table and the
right versions. Check in the source code the MySQL scripts who create
the
Hi Thomas,
That is a good question - right now the fix is dealing with the effect,
not solving the cause.I did a fast fixup to prevent any more crashes
for Brad.
But I still have to dig for the root cause of the problem.
Regards,
Bogdan
Thomas Gelf wrote:
Hi Bogdan,
do you have any
Hi Pascal,
I don't know why would that happen. I keep testing it and didn't see
this case..
Can you please raise the debug level and if you see it send me the part
of the log corresponding to the processing of the Subscribe and
generation of Notify.
Regards,
Anca
Pascal Maugeri wrote:
Hi
Hi Flavio,
Thanks for your response. I had to implement the similar scenario as
mentioned by you. I tried with different scenario by using forums,but in
vain. can you send the configuration implemented by your for local and pstn.
we have to authenticate our customers for certain destinations.
Thank you for your answer and suggestion Anca.To avoid to loosing time I
believe I should upgrade to 1.6 first (we're using 1.5 here). Then we'll try
to reproduce the issue and I'll come back to you with the log you asked.
Cheers
Pascal
On Thu, Oct 22, 2009 at 5:00 PM, Anca Vamanu
Iñaki Baz Castillo wrote:
You could also drop the second 200 in OpenSIPS by checking in reply_route[0]
check_trans(). It will return false for the second 200 OK as the
transaction
was removed upon recepit of the first 200. So the call drop(). However it
solves nothing since AST2 remains
Iñaki Baz Castillo wrote:
El Jueves, 22 de Octubre de 2009, Uwe Kastens escribió:
Hi,
After reading lots of docs and mailing lists, it looks like there is now
solution for asterisk available and looks like that might be a long way
till then.
If you mean the link below take into
Personally,I think broken UACs should behave like broken UACs. If you start
making exceptions, then they don't get fixed.
Things that are rigged to make them appear to work turn into problems that
are hard to detect. or become ignored until they become larger problems.
Like Bogdan said, just
Hi,
From my point of view I have no option but to solve that problem. If you
look at this special situation there is no solution to solve it with
asterisk without massive rewriting the code - or just hacking it in.
So yes from my point of view I would like to have that feature in TM and
it might
I am not sure if I have just misunderstood how serialize_branches() works or
if this is some kind of bug.
I have a registered user that gets called and I am using append_branch() to
add on that users mobile number. I am changing the Q values for both
branches to not be the same value, but when
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