I have just followed the thread for setting up failed calls in Radius
Mysetup is
Debian
Opensips 1.5.1 ( SVN ) i need to update to 1.5.2 later this weekend
Freeradius 2.1.6 with the patch mentioned by Mr NormB ( patch he was
mentioned in the patebin not worked )
I have manually edited the
Hi Brett,
You mean an PV returning the list with all the available codecs ?
Regards,
Bogdan
Brett Nemeroff wrote:
Is there anyway to write to an AVP the negotiated codec? That'd be
good for CDR purposes. Would I need a bunch of codec_exists in the
on_reply route checking for 200 OK?
On
2009/7/28 James Lamanna jlama...@gmail.com:
Hi,
I have some SPA942 and 962 phones that I'm trying to get BLF to work
properly with.
I've found it works correctly most of the time, however on occasion,
the BLF lights will get stuck as RED
(someone on a call) even though that person has hung
2009/7/23 andrei dragus andreidra...@yahoo.com:
Hello,
Methods have been added for SDP codec manipulation in the textops module.
Please update your module if you wish to use them.
There are 4 methods:
codec_exists(name[,clock]); //test if a codec exists
codec_delete(name[,clock]);
Hi Thiago,
I fixed the problem in the trunk version of OpenXCAP. You are probably
getting on error when OpenXCAP tries to communicate with OpenSIPS via
XMLRPC, but the error you are seeing hides the real one. You should now
be able to see that one.
Please take OpenXCAP from the darcs repository
Hi James,
What OpenSIPS version are you using?
Anca
James Lamanna wrote:
Hi,
I have some SPA942 and 962 phones that I'm trying to get BLF to work
properly with.
I've found it works correctly most of the time, however on occasion,
the BLF lights will get stuck as RED
(someone on a call)
Hi Andrew,
could you post the full trace of the call (SIP flaow) ? because in the
transfer scenario, there are multiple calls involved, so BYE may be part
of the scenario or some bogus reaction of a device.
Regards,
Bogdan
Andrew Yager wrote:
Hi Bogdan,
We do see a BYE at signalling level,
Iñaki,
From a purely philosophical perspective, I have nothing to add. I don¹t
know.
From an application perspective, this is exactly what we¹ve been waiting
for. Even if it doesn¹t function in all cases for everyone, it will
function in most portions of our network where only G.711 and G.729
Hi,
When running the MediaProxy ./setup build I get the following error:
---
./setup.py build
Traceback (most recent call last):
File ./setup.py, line 7, in ?
import mediaproxy
File
Hi Ross,
On 28 Jul 2009, at 12:53, Ross Beer wrote:
Hi,
When running the MediaProxy ./setup build I get the following error:
---
./setup.py build
Traceback (most recent call last):
File ./setup.py, line 7, in ?
import mediaproxy
Luci,
Hmm. If I don't load module 'pua_mi.so', the problem is:
Jul 28 09:22:03 debian openxcap[11488]: Starting factory
twisted.web.xmlrpc._QueryFactory instance at 0x887c64c
Jul 28 09:22:03 debian /usr/local/sbin/opensips[11559]:
ERROR:mi_xmlrpc:default_method: command pua_publish is not
Hi Jeff,
I looked overt the trace you sent me and the problem is the not the RR
param (as suspected), but the From URI which is changing across the
dialog. If you look into the trace, you will noticed that the domain
part (an IP actually) of the FROM header at INVITE time is different
than
I make Opensips Control System using a framework MVC to programming
(Catalyst/Perl), ORM for interface with database (DBIx::Class) and a
framework for JS (Jquery).
* Manager subscribers, domain and pdt tables.
* See active_watchers, presentity, watchers, xcap tables. (all fields
and XML
Hi Bogdan,
I think I see what you're referring to. The domain of the From field at
INVITE time is not the same as the domain of the To field in the BYE from
the upstream proxy with the PSTN gateway behind it, correct?
I verified it's the PSTN gateway changing the domain, not the upstream
proxy.
Yes Jeff, this is correct.
Regards,
Bogdan
Jeff Pyle wrote:
Hi Bogdan,
I think I see what you're referring to. The domain of the From field at
INVITE time is not the same as the domain of the To field in the BYE from
the upstream proxy with the PSTN gateway behind it, correct?
I verified
Got it.
I contacted my provider, citing the third paragraph of section 12.2.1.1 of
RFC 3261 where it talks about keeping the From and To URIs the same for
compatibility with RFC 2543. We'll see what they say. That section goes on
to say this requirement will likely be removed in the future,
Hi Iñaki,
Iñaki Baz Castillo wrote:
2009/7/23 andrei dragus andreidra...@yahoo.com:
Hello,
Methods have been added for SDP codec manipulation in the textops module.
Please update your module if you wish to use them.
There are 4 methods:
codec_exists(name[,clock]); //test if a codec
i mean,i hope one sipid can be log on by one person in my opensips at a
time.the other one cant log on successful even if have same id and password.
then how to limit it.
Worm regard
___
好玩贺卡等你发,邮箱贺卡全新上线!
I don't really need codec manipulation so much as just knowing what codec
was used (yes in a PV). Not a list of available codecs, but which codec was
negotiated and used. I don't know SDP very well so I'm not sure if that's
immediately discernible. This example was given earlier. Lets say I want
28 jul 2009 kl. 16.41 skrev Brett Nemeroff:
I don't really need codec manipulation so much as just knowing what
codec was used (yes in a PV). Not a list of available codecs, but
which codec was negotiated and used. I don't know SDP very well so
I'm not sure if that's immediately
Olle E. Johansson wrote:
As far as I know, there's no way in SIP you can determine what codec
actually was used if the offer/answer resultet in multiple codecs.
I was just going to say that. Even if you mimic the exact algorithm
used by the offer and answer side, since there is no
It's worth pointing out that no member of the OpenSER project stack has
been a pure SIP proxy for very long; they have certain UAS features
(registrar, PUA, NAT ping, etc.) As Bogdan said, a pure proxy would not
be terribly useful in most scenarios in which the project is deployed.
--
Alex
28 jul 2009 kl. 16.53 skrev Alex Balashov:
It's worth pointing out that no member of the OpenSER project stack
has
been a pure SIP proxy for very long; they have certain UAS features
(registrar, PUA, NAT ping, etc.) As Bogdan said, a pure proxy would
not
be terribly useful in most
2009/7/28 Thiago Rondon thi...@aware.com.br:
I make Opensips Control System using a framework MVC to programming
(Catalyst/Perl), ORM for interface with database (DBIx::Class) and a
framework for JS (Jquery).
* Manager subscribers, domain and pdt tables.
* See active_watchers, presentity,
Hi Brett,
as you probably already found out, you cannot (as proxy) say what codec
is used in a call. You (as a proxy) see just the offerings (what codecs
are available on each side) - after that, each side is free to select
and use one of the offered codecs - so, only inspecting the RTP stream
All,I was reading the thread regarding the uac_replace_from issues Jeff
brought up and was thinking my issue may be similar.
I have a carrier who sends me BYE messages with a RURI that does NOT match
the Contact header in the 200 OK. Of course, OpenSIPs replies with a 404 Not
Here.
The last
Hi Bogdan,
On 7/28/09 11:46 AM, Bogdan-Andrei Iancu bog...@voice-system.ro wrote:
Hi Jeff,
I agree that the RFC3261 keeps this requirement only for backcompat
reasons. unfortunately, if we cannot rely on the uri preserving, we will
have to store both the old and new uri in the message so
Jeff Pyle wrote:
Options 2 is to to use dialog module as support for locally storing the
changed URI, but it will create the dependency to the dialog module.
This seems like a viable option, as long as these values could be stored in
the database with the proper dbmode set on the dialog
Hi Eason,
yes, the sip:opensips.mydomain.com is your RURI - and probalby your
opensips does not recognize the opensips.mydomain.com domain as a domain
to locally process. Have you configured it somewhere?
Regards,
Bogdan
hsuan wrote:
Dear Bogdan,
Thank you for your feedback again...
The
Hi Andrew,
Andrew Yager wrote:
Hi,
We are running opensips 1.5.2-notls, and have usrloc working well in
theory.
We are trying to set up replication to a second opensips server, with
one of two methods:
* DB Replication only (usrloc db_mode 3) OR
* t_replicate (with usrloc db_mode 2)
Hi Brett,
The dialog module can be used to validate the sequential requests (based
on the stored info like RR and contacts) - maybe some functions to do
that will be useful :). Thinking in the future (but debatable), you can
do fixing of the sequential requests (based on the stored info).
Hi Thiago,
This looks interesting , especially the user management part ;).
Please list it in the opensips related software section -
http://www.opensips.org/Resources/RelatedSoftware
Thanks and regards,
Bogdan
Thiago Rondon wrote:
I make Opensips Control System using a framework MVC to
I am using 1.5.2
--James
On Jul 28, 2009, at 1:48, Anca Vamanu a...@opensips.org wrote:
Hi James,
What OpenSIPS version are you using?
Anca
James Lamanna wrote:
Hi,
I have some SPA942 and 962 phones that I'm trying to get BLF to work
properly with.
I've found it works correctly most
Hi,
I have compiled and installed mediaproxy sucessfuly however when starting it
says 'fatal error: need python-gnutls version 1.1.8 or higer but only 1.1.6 is
installed' however I have installed 1.1.9 and it still says the same.
Is there a sys-link I need to update somewhere to point
Most likely you have multiple installed versions and it picks the
wrong one (the older). You can verify what it finds by default and
where it is located by running this:
python -c import sys, gnutls; print gnutls.__version__; print
sys.modules['gnutls']
On 28 Jul 2009, at 21:04, Ross Beer
Hi,
That has give me:
module 'gnutls' from '/var/lib/python-support/python2.6/gnutls/__init__.pyc'
Which is the correct python version, however how do I get this python-gnutls to
update?
Do I need to set a variable when installing python-gnutls from source?
Any advice is
On Tuesday 28 July 2009, Ross Beer wrote:
That has give me:
module 'gnutls' from '/var/lib/python-
support/python2.6/gnutls/__init__.pyc'
Which is the correct python version, however how do I get this python-gnutls
to update?
It depends how you installed it in the first place.
El Martes, 28 de Julio de 2009, Alex Balashov escribió:
It's worth pointing out that no member of the OpenSER project stack has
been a pure SIP proxy for very long; they have certain UAS features
(registrar, PUA, NAT ping, etc.) As Bogdan said, a pure proxy would not
be terribly useful in
I can confirm that problems with off-hook or rather early state are
reproducible on OpensipS bran 1.5 Rev 5916.
There is Reply 412 Conditional request failed for PUBLISH that has
stateearly/state in message body.
This happens only when call is made by watched user to SPA942 that is
watching that
Inaki,
Why do you complain about having features in a software that you did
not write just because you do not see a personally use in them?
Adrian
On Jul 28, 2009, at 10:19 PM, Iñaki Baz Castillo wrote:
El Martes, 28 de Julio de 2009, Alex Balashov escribió:
It's worth pointing out that no
El Martes, 28 de Julio de 2009, Adrian Georgescu escribió:
Inaki,
Why do you complain about having features in a software that you did
not write just because you do not see a personally use in them?
Adrian, I don't complain, please don't misunderstand me.
I just say that many new features
El Martes, 28 de Julio de 2009, Adrian Georgescu escribió:
Making arbitrary judgements about others people work just for the sake
of not agreeing with them is not adding any value, is it?
Ok, if you prefer I would say that, IMHO, such new features would fit better
on top of a real b2bua
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