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 modul
Making arbitrary judgements about others people work just for the sake
of not agreeing with them is not adding any value, is it?
Adrian
On Jul 28, 2009, at 10:51 PM, Iñaki Baz Castillo wrote:
> El Martes, 28 de Julio de 2009, Adrian Georgescu escribió:
>> Inaki,
>>
>> Why do you complain about
I know no one is asking my opinion here... :) but as long as these features
are well isolated in their own modules I say the more features the merrier.
Just don't go hacking up the core to support features that were never really
intended to support. We've all seen what THAT does to these projects.
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
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
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
early in message body.
This happens only when call is made by watched user to SPA942 that is
watching that user.
I ha
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
On Tuesday 28 July 2009, Ross Beer wrote:
> That has give me:
>
>
>
>
>
>
>
> 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. You mentioned you
installed both 1.1.6 and 1.1.9. How did you do t
Hi,
That has give me:
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 much appreciated.
Ross
> CC: r...@ag-projects.com; users@lists.opensips.or
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,
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 py
I am using 1.5.2
--James
On Jul 28, 2009, at 1:48, Anca Vamanu 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 of
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 pr
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).
Rega
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 d
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...
>
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 th
Hi Bogdan,
On 7/28/09 11:46 AM, "Bogdan-Andrei Iancu" 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 that we can
> re
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 messa
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 that we can
restore it when coming from both up and down stream directions. This can
be
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
2009/7/28 Thiago Rondon :
>
> 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
On Tue, Jul 28, 2009 at 8:04 PM, 星宇 刘 wrote:
> 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.
>
>
Hi
have looked this registrar module
http://www.opensips.org/h
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
28 jul 2009 kl. 16.51 skrev Alex Balashov:
> 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
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
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 knowl
Yeah, I was afraid of that. perhaps what would have to be done is to limit
to force one codec.. seems messy :/
So not sure if there would be a way to limit the number of paths for a given
codec..
On Tue, Jul 28, 2009 at 9:47 AM, Olle E. Johansson wrote:
>
> 28 jul 2009 kl. 16.41 skrev Brett Nem
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 dis
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 to
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
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2009/7/28 Bogdan-Andrei Iancu :
> The codec related ops are very "light" and suitable for a proxy, in the way
> that the proxy influence the codec negotiations, but without bringing the
> SDP negotiation into an inconsistent state. This is the reason why there is
> no "add_codec" operation (even if
Hi Iñaki,
Iñaki Baz Castillo wrote:
> 2009/7/23 andrei dragus :
>
>> 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
>
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, usin
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 ve
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.
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 docume
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 the
Luci,
Hmm. If I don't load module 'pua_mi.so', the problem is:
Jul 28 09:22:03 debian openxcap[11488]: Starting factory
Jul 28 09:22:03 debian /usr/local/sbin/opensips[11559]:
ERROR:mi_xmlrpc:default_method: command pua_publish is not available!
Jul 28 09:22:03 debian openxcap[11488]: error:
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 media
2009/7/28 Ross Beer :
> Can anyone offer any advice, I have installed all of the requirments. I
> followed instructions from http://www.smartvox.co.uk
Do you use Debian Unstable?
--
Iñaki Baz Castillo
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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 "/home/voicehost/MediaProxy/mediaproxy-2.3.5/medi
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
I was waiting for a reason to try 1.6... :)
On 7/28/09 3:53 AM, "Bogdan-Andrei Iancu" wrote:
> Hi Jeff,
>
> Only in 1.6, as this is a new piece of code and only fixes are
> backported to stable 1.5.
>
> Regards,
> Bogdan
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U
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 leve
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 cal
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
2009/7/23 andrei dragus :
>
> 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]); //delete a codec
>
2009/7/28 James Lamanna :
> 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 up.
When I te
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?
>
>
Hi Jeff,
Only in 1.6, as this is a new piece of code and only fixes are
backported to stable 1.5.
Regards,
Bogdan
Jeff Pyle wrote:
> Rockstar! Is this available in 1.5 as well, or just 1.6?
>
>
> - Jeff
>
>
>
> On 7/23/09 5:46 AM, "andrei dragus" wrote:
>
>
>> Hello,
>>
>> Methods have bee
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 file
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