We did, it f'd up in the long run.
-----Original Message-----
From: Juha Heinanen [mailto:jh@tutpro.com]
Sent: Friday, June 17, 2005 1:10 PM
To: Matt Schulte
Cc: users(a)openser.org
Subject: [Users] extcmd removed?
Matt Schulte writes:
> leaks/crashes, there is no alternativer other than writing a module.
Of > which no one in our company can do so :-)
hire someone from outside to write your module.
-- juha
I have successfully installed ser and mediaproxy on my network, it is a /27.
Now to load balance I installed a second mediaproxy server on a datacenter, so on another network.
I configure dns and the mediaproxy.ini (on the new server) to work with the new ip address.
I disbaled all firewall rules to test the proxy.
But it didn't work, the call is established but no rtp flow between the ua and mediaproxy.
This is the debug from the dispacher:
Jun 17 18:10:32 voip proxydispatcher[26275]: command request 493086ee-17262f1a(a)10.254.0.42 10.254.0.42:16440:audio 213.178.216.33 voadsl.italycomnet.it local 213.178.208.130 remote Sipura/SPA2100-3.1.3 info=from:prova@voadsl.italycomnet.it,to:0828720607@voadsl.italycomnet.it,fromtag:56eac4de28c789eao0,totag:
Jun 17 18:10:32 voip proxydispatcher[26275]: forwarding to mediaproxy on voicegw-mi.italycomnet.it:25060: got: '213.178.208.130 35006'
Jun 17 18:10:32 voip proxydispatcher[26275]: command execution time: 289.48 ms
Jun 17 18:10:32 voip proxydispatcher[26275]: command lookup 493086ee-17262f1a(a)10.254.0.42 213.178.208.130:13308:audio 213.178.208.130 voadsl.italycomnet.it local voadsl.italycomnet.it unknown Asterisk=20Italycom=20Mix info=from:prova@voadsl.italycomnet.it,to:0828720607@voadsl.italycomnet.it,fromtag:56eac4de28c789eao0,totag:as58d31ab2
Jun 17 18:10:33 voip proxydispatcher[26275]: forwarding to mediaproxy on voicegw-mi.italycomnet.it:25060: got: '213.178.208.130 35006'
Jun 17 18:10:33 voip proxydispatcher[26275]: command execution time: 183.21 ms
Jun 17 18:10:40 voip proxydispatcher[26275]: command lookup 493086ee-17262f1a(a)10.254.0.42 213.178.208.130:13308:audio 213.178.208.130 voadsl.italycomnet.it local voadsl.italycomnet.it unknown Asterisk=20Italycom=20Mix info=from:prova@voadsl.italycomnet.it,to:0828720607@voadsl.italycomnet.it,fromtag:56eac4de28c789eao0,totag:as58d31ab2
Jun 17 18:10:41 voip proxydispatcher[26275]: forwarding to mediaproxy on voicegw-mi.italycomnet.it:25060: got: '213.178.208.130 35006'
Jun 17 18:10:41 voip proxydispatcher[26275]: command execution time: 175.07 ms
Jun 17 18:10:46 voip proxydispatcher[26275]: command delete 493086ee-17262f1a(a)10.254.0.42 info=
Jun 17 18:10:46 voip proxydispatcher[26275]: forwarding to mediaproxy on voicegw-mi.italycomnet.it:25060: got: ''
Jun 17 18:10:46 voip proxydispatcher[26275]: command execution time: 191.27 ms
this is the debug from the mediaproxy server:
Jun 17 18:10:32 voicegw-mi mediaproxy[4213]: command request 493086ee-17262f1a(a)10.254.0.42 10.254.0.42:16440:audio 213.178.216.33 voadsl.italycomnet.it local 213.178.208.130 remote Sipura/SPA2100-3.1.3 info=from:prova@voadsl.italycomnet.it,to:0828720607@voadsl.italycomnet.it,fromtag:56eac4de28c789eao0,totag:,dispatcher
Jun 17 18:10:32 voicegw-mi mediaproxy[4213]: session 493086ee-17262f1a(a)10.254.0.42: started. listening on 213.178.208.130:35006
Jun 17 18:10:32 voicegw-mi mediaproxy[4213]: command execution time: 1.61 ms
Jun 17 18:10:33 voicegw-mi mediaproxy[4213]: command lookup 493086ee-17262f1a(a)10.254.0.42 213.178.208.130:13308:audio 213.178.208.130 voadsl.italycomnet.it local voadsl.italycomnet.it unknown Asterisk=20Italycom=20Mix info=from:prova@voadsl.italycomnet.it,to:0828720607@voadsl.italycomnet.it,fromtag:56eac4de28c789eao0,totag:as58d31ab2,dispatcher
Jun 17 18:10:33 voicegw-mi mediaproxy[4213]: command execution time: 0.56 ms
Jun 17 18:10:33 voicegw-mi mediaproxy[4213]: session 493086ee-17262f1a(a)10.254.0.42: called signed in from 213.178.208.130:13308 (RTP) (will return to 213.178.208.130:13308)
Jun 17 18:10:33 voicegw-mi mediaproxy[4213]: warning: Received packet from a third party: 213.178.216.2:16440
Jun 17 18:10:40 voicegw-mi mediaproxy[4213]: command lookup 493086ee-17262f1a(a)10.254.0.42 213.178.208.130:13308:audio 213.178.208.130 voadsl.italycomnet.it local voadsl.italycomnet.it unknown Asterisk=20Italycom=20Mix info=from:prova@voadsl.italycomnet.it,to:0828720607@voadsl.italycomnet.it,fromtag:56eac4de28c789eao0,totag:as58d31ab2,dispatcher
Jun 17 18:10:40 voicegw-mi mediaproxy[4213]: command execution time: 0.18 ms
Jun 17 18:10:46 voicegw-mi mediaproxy[4213]: command delete 493086ee-17262f1a(a)10.254.0.42 info=dispatcher
Jun 17 18:10:46 voicegw-mi mediaproxy[4213]: session 493086ee-17262f1a(a)10.254.0.42: 0/646/0 packets, 0/38760/0 bytes (caller/called/relayed)
Jun 17 18:10:46 voicegw-mi mediaproxy[4213]: session 493086ee-17262f1a(a)10.254.0.42: ended.
Jun 17 18:10:46 voicegw-mi mediaproxy[4213]: command execution time: 0.42 ms
Jun 17 18:11:52 voicegw-mi sshd(pam_unix)[4228]: session opened for user root by root(uid=0)
Seems that is not able to find the right pubblic address of the ua....
Any help is appreciated.
Thanks
Rosario
Hi folks,
thanks to Peter and Cesc we have TLS support buit-in OpenSER. The
implementation is only on CVS head (development branch).
To getting do:
- if you already have a local checkout of CVS head:
cvs update -dP
- no local checkout of CVS head:
cvs
-d:pserver:anonymous@cvs.sourceforge.net:/cvsroot/openser login
cvs -z3
-d:pserver:anonymous@cvs.sourceforge.net:/cvsroot/openser co -P sip-server
Full documentation about how to compile and use it (script example
included) can be found in the documentation repository:
http://openser.org/docs/tls.html
The next step will be to convince as many people as possible to use it
and to test it for eventual bugs/problems. I will be very helpful in
moving towards a stable version.
regards,
Bogdan
Hello,
I have configured SER and asterisk to allow me to make calls to the PSTN
network, however on my voip phone (behind NAT) I am having issues with the
voip tx audio dropping out after 30 seconds. Now I'm guessing it's a nat
issue but even that doesn't really make sense!!
Why, well because the only the TX of the voip phone drops out (ie the PSTN
phone cannot hear what is said on the voip phone). The PSTN phone can still
transmit audio to the voip phone (through the nat).
Anyway in SER, I have set the natping_interval to 5 seconds, and this still
doesn't resolve the issue. Strangely at the time that the audio disconnects
Asterisk is sending my phone an INVITE message. Why would it do this mid
call?
I'm using SER0.9.0+Asterisk as my platform.
Any pointers??
JB
Dear Bogdan, Dear Daniel,
As a user of SER for number of years now I really appreciate your work and effort to liberate SER from the
clutches of iptelorg with its microsoft like mentality.
I am sure it was not easy for you to stand-up against those who have paid your salary for a number of years and have given you the knowledge and software with which you have built your own company.
I also agree with you that the Fraunhofer as the maintainer of the iptel.org and
berlios sites has in no way the right to refer on these sites
to a spinn-off in which it is involved in -having
invested a huge sum in this project including your salary in the
last years is surely no excuse.
I am also of your opinion that the benefits of the users should
be the utmost goal of all who are involved in SER. Therefore my
question is: will the next release of openSER also include the VNT and VPS products
your company are offering? Providing these components to the community will
surely help the SER users to offer better VoIP services -which as
you have stressed should be our major goal. I would recommend to publish your products under the BSD license so that we could include them into our own products as well? We are building an NGN product that would benefit greatly from your components. iptelorg has actually asked us to pay for their source code. Those guys have surely not realized yet that SER does not belong to them but to the users and community.
Best regards
--
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I believe the community should be allowed to tell the core developers their opinion and I have thus created a poll at http://onsip.org/
(http://www.onsip.org/modules/xoopspoll/index.php?poll_id=5)
I apologize that you will have to log in (or register if you have not yet done so) in order to vote. This is how Xoops (the content management system) avoids voting manipulation through (otherwise allowed) multiple votes from one user.
Below are the statements that you can agree to or not. Please send me a private note if you have suggestions for other statements that will be informative.
Regards,
Greger
-------------------------------------------------------------------------------------
What is your reaction to the OpenSER announcement?
Overall, a separate OpenSER will benefit the users
Overall, a separate OpenSER will not benefit the users
I will go for the "old" SER
I will go for the new OpenSER
The conflict should be resolved into one project
Splitting in two is a good thing
Hi bogdan,
Need I write script for failure_route and
onreply_route in ser.cfg to support parallel fork or I
only lets openser to do that by itself.
If need, how to write the script?
By the way, how to support serial fork?
Regards,
Jimway
--- Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>
wrote:
> Hi Jimmy.
>
> see inline comments. As short note, what you are
> describing in parallel
> forking and it behaves exactly as you thing (see RFC
> 3261).
>
> regards,
> bogdan
>
> jimmy way wrote:
>
> >Hi all,
> >
> > The question is:
> > 1. a user A registar from multi address, like
> > UA1 registrar from address1
> > UA2 registrar from address2
> > UA3 registrar from address3
> > and so on
> > so in db "location" there are multi address
> in
> >the list about user A
> > 2. if set append_branch as on in registrar
> module,
> >when user B invite user A, Openser server will
> append
> >all these address in the invite packet. So all
> these
> >user will get the call. normally all UAs ringing,
> then
> > 3. if one of these UAs accept the call, how
> about
> >other UAs, will they stop ringing?
> >
> >
> yes, the other pending branches will be
> automatically cancelled.
>
> > 4. if one of these UAs deny(40x) the call, how
> >about other UAs, will the still ringing?
> >
> >
> yes - there are two way for a parallel forked call
> to end: either one of
> the branches picks up (2xx reply), either all of
> them send negative
> replies (>=300).
>
> > 5. if one of these UAs not connected, how about
> >caller the callee do?
> >
> >
> if one of the branches is not connected (but still
> registered), an
> internal timeout (408) will be generated for it.
>
> >
> >tks.
> >Jimway
> >
> >
> >
>
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hi.
as I make test connection beetwen kphone ( sip client ) and me rtpproxy.
thanks for your time.
--
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Bruno "Niggas" Oliveira
Belo Horizonte - MG
Msn: n1gg4s(a)gmail.com
Icq: 176314647
"Todo o nosso descontentamento por aquilo
que nos falta procede da nossa falta de
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