Re: [OpenSIPS-Users] OpenSIPS 1.6 on Ubuntu

2011-02-05 Thread Pradeep Patil
Hi,

But I am not able to do so:

When I try to edit the file getting follwoing error: W10: Warning: Changing
a readonly file 

And file bot getting saved.

On Fri, Feb 4, 2011 at 11:28 PM, Grygoriy Dobrovolskyy
megaho...@gmail.comwrote:

 Hi Bogdan, you changed the mic once when i asked you, after the audio
 was perfect. Do you remember ?

 2011/2/4 Bogdan-Andrei Iancu bog...@opensips.org:
  Hi Tyler,
 
  Unfortunately it is not so simply as mic problem :) - the recording was a
  done from a different location than where I was (so, across the wild
  internet)
 
  I will try to re-register the webinar, just to have a good audio.
 
  Regards,
  Bogdan
 
  Tyler Merritt wrote:
 
  Dave,
 
  The audio on some of the webinars that I have watched has been almost
  unintelligible :(  I like webinars - I present many of them in my work
 for
  our customers, but I couldn't really hear well.
 
  I can't attend the live webinars as I'm in Tokyo - they happen at like 3
  am.
 
  Anyway to clean up the audio?  Bogdan - can I send you a mic better mic
 :)
 
   http://pbxtra.fonality.com/products/hud/ *Tyler Merritt*. Sales
  Engineer.   Contact: tmerr...@fonality.com mailto:ty...@fonality.com
 |
  310.861.4300 x 8850 |   fonality.com http://www.fonality.com | SE
 Blog
  http://fonalityse.wordpress.com/   http://www.twitter.com/fonality
   http://www.linkedin.com/pub/fonality-inc/15/a2b/13b
   http://www.facebook.com/Fonality  
 http://www.youtube.com/user/Fonality
  http://feeds.feedburner.com/fonalitypressreleases
  http://www.trixbox.org
 
 
 
 
  On Thu, Feb 3, 2011 at 2:40 PM, Dave Singer dave.sin...@wideideas.com
  mailto:dave.sin...@wideideas.com wrote:
 
 The best place to start is http://www.opensips.org/
 In the left column of the web page there is a section titled
 Resources
 with links to many very helpful resources. Your using the mailing
 list
 so you probably already have seen them to get here.
 So. Where are you getting stuck? We need specifics in order to
 help out.
 
 Also when you have a question you should start your own thread and
 not
 use an existing thread unless it is completely relevant to what your
 asking/stating.
 
 FYI: The webinars are VERY important for getting an understanding of
 how the whole thing works. With SIP the big picture is very
 important!
 With out them you'll learn a lot of things the hard way like I did
 before they were available.
 Another good way to learn is to follow the mailing list discussions.
 
 Welcome to the club,  ;-)
 Dave
 
 P.S. The software, documentation, mailing list, IRC, etc are all free
 resources. The people helping you out are not getting paid to do it.
 So an attitude of appreciation with patience will get you the best
 millage. If you need more support there are those willing to do
 contract support. See http://www.opensips.org/Resources/Business
 
 On Wed, Feb 2, 2011 at 7:16 PM, Pradeep Patil
 pradeep.pati...@gmail.com mailto:pradeep.pati...@gmail.com
 wrote:
  Anyone can help please in installing Opensip on Ubuntu.
 
 
 
 
 
  On Wed, Feb 2, 2011 at 1:15 AM, Duane Larson
 duane.lar...@gmail.com mailto:duane.lar...@gmail.com wrote:
 
  The first thing you should do is
  http://www.packtpub.com/article/installation-of-opensips-1.6
 
  You can watch the webinars here
  http://www.opensips.org/Resources/Webinars
 
  You should join the mailing list
  http://www.opensips.org/Resources/MailingLists
 
  To search old mailing list posts I use
 
 
 
 
 http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html
 
  Sounds like what you need to do is to actually create a
 user/subscriber so
  that opensips can register the x-lite client.  For that you
 need to use the
  opensipsctl command or the osipsconsole.
 
 
 
  On Tue, Feb 1, 2011 at 12:21 PM, Robin Malhotra
 rocky...@gmail.com mailto:rocky...@gmail.com
  wrote:
 
  Guys I a newbie to OpenSIPS
 
   I have installed opensips and mysql on ubuntu following some
  instructions. I have also installed x-lite. Now how to
 register a user in
  opensips and to use it with the client ? I am stuck, please
 let me know
  Regards
  Ricky
 
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  --
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  Duane
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Re: [OpenSIPS-Users] Change in SDP/RTP routing with Opensips 1.4 and 1.6

2011-02-05 Thread Ovidiu Sas
Have you checked if the your config is altering the SDP?
Check the SDP for the same message before coming to your opensips
server and after leaving the opensips server.
If the SDP is altered and the IP in SDP is pointing to your opensips
server, then there is your problem.


Regards,
Ovidiu Sas

On Fri, Feb 4, 2011 at 7:53 PM, Chris Stone axi...@gmail.com wrote:
 Hello - We have an Opensips 1.4 server that routes incoming calls to a
 couple of different Asterisk servers and to upstream providers. All
 working great and with the current config, the Opensips server only
 handles the SIP traffic - all of the audio is between the UAs and
 Asterisk servers.

 Am building another Opensips server and decided to do it with the
 1.6.3 release. With virtually the same config (really only had to
 change a couple of things at the top like loading signal.so, dropping
 the loading of xlog.so, etc), now Opensips is in the picture for all
 of the audio as well as SIP traffic. Was troubleshooting an issue with
 no audio in either direction when calling in - so was capturing
 traffic on the Opensips server, saw the SIP traffic, call relayed
 correctly to the Asterisk server where and IVR was played and all the
 audio was going back to the Opensips server. Did the same test on the
 Opensips 1.4 server and no audio packets - which is what I want.

 Done a days searching and am not finding the fix. Someone know what
 changed and how I can get the same behavior I have with 1.4 with the
 1.6 release?


 Thanks!



 Chris

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Re: [OpenSIPS-Users] [SR-Users] authentication is not working

2011-02-05 Thread Danny Dias
2011/2/4 Iñaki Baz Castillo i...@aliax.net

 2011/2/4 Danny Dias ing.diasda...@gmail.com:
   different emails...what's the problem? i have a doubt and then i
   ask...thats
   the purpose of the forum, if you are not going to help, don't
 answer...
 
  You are showing same config file (just replaced kamailio with
  opensips) and exactly same SIP traces. So please clarify if you are
  using Kamailio or OpenSIPS since they are different.
 
  oooh...different? both are proxys or not?

 YXA [*] is also a SIP proxy (written in Erlang language). Would a
 question about YXA make sense in this maillist just because YXA is a
 SIP proxy? :)

 [*] http://www.stacken.kth.se/project/yxa/



Nop, you're right.


  now i'm lost...it's the same
  trace, it happened with OpenSIPS, i'm using Kamailio and Opensips, but
 the
  problem was with Opensips (not problem of the application, the problem
 was
  in my hands, bad configuration, etc...), as i said before, i wont ask the
  same questions to different lists again...but i thought that they would
 have
  the exactly behavior in my problem...

 Maybe, but the recipes you will get in each mailist (so different
 projects) could be different. This is, somebody in this maillist could
 try to help you but what he suggests could be invalid for OpenSIPS
 (newest versions of both projects differ much more than their 1.5.X
 version).


ok




  If you are using Kamailio and report a problem in OpenSIPS maillist
  (or vice versa) then you are providing not valid information (both
  projects are not the same, even more in newer versions).
 
  So what are you using and which exact version?
 
  For Kamailio 1.5 and for OpenSIPS 1.6.3 (right now Just for testing
  scenarios...)

 But the reported problem occurs with your OpenSIPS installation, and
 provided configuration and SIP traces are taken using OpenSIP, am I
 right?


yep...


 If so, I strongly suggest you to just ask in OpenSIPS maillist due to
 the explained above. The help you'll get there would be more valid
 for you.

 Cheers.



cool!


 --
 Iñaki Baz Castillo
 i...@aliax.net

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[OpenSIPS-Users] Topology hiding - B2B_LOGIC

2011-02-05 Thread Maciej Bylica
Hi.

I am running Opensips 1.6.3 and trying to do topology hiding.
This is my scenario:Operator_1 --  my Opensips -- Operator_2
The goal is not to convey any information of Operator_2 to Operator_1
like Contact, User-Agent headers and so on and to do rtp proxying.
For rtp proxying i've installed rtpproxy and it works fine.
But still the question is about signalization and SDP (o= part)
I ran through a few posts and found out that the answer is B2B
functionality here - so B2B_LOGIC.

Are there any other wayouts or this is the only way i may follow.

Thanks in advance,
Maciej.

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