Re: [OpenSIPS-Users] Force media through the Opensips server

[TeleCube - John]

Hi Adam,

Thanks, the fog is starting to lift now...  :-)

I'll make some changes and come back with results shortly.

Cheers,
John

Adam Twardowski wrote:

John,

The o= is the Origin IP, which doesn't really matter, the c= is the 
connection IP, which is the important one.  You probably want to use 
force_rtp_proxy() to rewrite the sdp and make it go through the 
rtpproxy, and in your asterisk SIP peer, you can either set the peer 
IP to 4.5.6.7 and have your proxy re-write the RURI to 1.2.3.4 and 
relay it along, or you could set outboundproxy=4.5.6.7 in sip.conf on 
your sip peer and just have opensips relay it.  You may also need to 
setup an on-reply route to capture 180/183 messages with SDP and 
rewrite the IP's in there too.


--Adam

On Fri, Aug 27, 2010 at 10:45 PM, TeleCube - John 
mailto:j...@telecube.com.au>> wrote:


Hi Max,

Thanks for your reply.

I have rtpproxy running and I think it's all configured ok.

I can use fix_nated_sdp() with flags and see the changes in the
sdp lines in the packets.

What I am struggling with is just how to go about telling the
asterisk server and the outbound route that all traffic is to run
through my proxy

I'm not sure what the purpose of the o= and c= lines are and at
which stage in the packet flow I need to re-write which lines.

If anyone can offer any pointers that will be awesome.

Kind regards,
John


Max Mühlbronner wrote:

  Hello,

Yes, the opensips will rewrite the sdp like you said, so it will contain 
the ip of the rtpproxy (or mediaproxy).  It can be used to proxy the rtp 
traffic to a specific destination.


But the rtpproxy does not necessarily have to be the same machine as 
your Opensips. The rtpproxy is a different tool which will be enabled by 
setting the relevant modparam settings and also calling the rtpproxy in 
your routing-script.


http://voiprookie.blogspot.com/2009/04/rtpproxy-12x-installation.html

maybe there are also other, better tutorials but this link seems to be a 
good start for setting up rtpproxy with opensips.



BR

Max M.


Am 27.08.2010 15:36, schrieb TeleCube - John:
  

Hi,

I have a setup as follows:

1.2.3.4 - Outbound Route
4.5.6.7 - Opensips proxy
4.5.6.8 - Asterisk server

The outbound route will only allow any traffic, signaling and media, via
the 4.5.6.7 IP address. It cannot currently accept media from a
different ip than the signaling.

Currently the astx server sends media direct to the outbound route.

Signaling is fine but I need to force the media stream from the asterisk
server through the opensips proxy.

Am I correct in expecting that nathelper/rtpproxy can do this?

Is it simply a matter of re-writing the sdp detail that will tell the
asterisk server to route media through the opensips ip address?

Any examples will be greatly appreciated.

Thanks,
John

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Re: [OpenSIPS-Users] Force media through the Opensips server

[Adam Twardowski]

John,

The o= is the Origin IP, which doesn't really matter, the c= is the
connection IP, which is the important one.  You probably want to use
force_rtp_proxy() to rewrite the sdp and make it go through the rtpproxy,
and in your asterisk SIP peer, you can either set the peer IP to 4.5.6.7 and
have your proxy re-write the RURI to 1.2.3.4 and relay it along, or you
could set outboundproxy=4.5.6.7 in sip.conf on your sip peer and just have
opensips relay it.  You may also need to setup an on-reply route to capture
180/183 messages with SDP and rewrite the IP's in there too.

--Adam

On Fri, Aug 27, 2010 at 10:45 PM, TeleCube - John wrote:

>  Hi Max,
>
> Thanks for your reply.
>
> I have rtpproxy running and I think it's all configured ok.
>
> I can use fix_nated_sdp() with flags and see the changes in the sdp lines
> in the packets.
>
> What I am struggling with is just how to go about telling the asterisk
> server and the outbound route that all traffic is to run through my proxy
>
> I'm not sure what the purpose of the o= and c= lines are and at which stage
> in the packet flow I need to re-write which lines.
>
> If anyone can offer any pointers that will be awesome.
>
> Kind regards,
> John
>
>
> Max Mühlbronner wrote:
>
>   Hello,
>
> Yes, the opensips will rewrite the sdp like you said, so it will contain
> the ip of the rtpproxy (or mediaproxy).  It can be used to proxy the rtp
> traffic to a specific destination.
>
> But the rtpproxy does not necessarily have to be the same machine as
> your Opensips. The rtpproxy is a different tool which will be enabled by
> setting the relevant modparam settings and also calling the rtpproxy in
> your routing-script.
> http://voiprookie.blogspot.com/2009/04/rtpproxy-12x-installation.html
>
> maybe there are also other, better tutorials but this link seems to be a
> good start for setting up rtpproxy with opensips.
>
>
> BR
>
> Max M.
>
>
> Am 27.08.2010 15:36, schrieb TeleCube - John:
>
>
>  Hi,
>
> I have a setup as follows:
>
> 1.2.3.4 - Outbound Route
> 4.5.6.7 - Opensips proxy
> 4.5.6.8 - Asterisk server
>
> The outbound route will only allow any traffic, signaling and media, via
> the 4.5.6.7 IP address. It cannot currently accept media from a
> different ip than the signaling.
>
> Currently the astx server sends media direct to the outbound route.
>
> Signaling is fine but I need to force the media stream from the asterisk
> server through the opensips proxy.
>
> Am I correct in expecting that nathelper/rtpproxy can do this?
>
> Is it simply a matter of re-writing the sdp detail that will tell the
> asterisk server to route media through the opensips ip address?
>
> Any examples will be greatly appreciated.
>
> Thanks,
> John
>
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>
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Re: [OpenSIPS-Users] Force media through the Opensips server

[TeleCube - John]

Hi Max,

Thanks for your reply.

I have rtpproxy running and I think it's all configured ok.

I can use fix_nated_sdp() with flags and see the changes in the sdp 
lines in the packets.


What I am struggling with is just how to go about telling the asterisk 
server and the outbound route that all traffic is to run through my proxy


I'm not sure what the purpose of the o= and c= lines are and at which 
stage in the packet flow I need to re-write which lines.


If anyone can offer any pointers that will be awesome.

Kind regards,
John

Max Mühlbronner wrote:

  Hello,

Yes, the opensips will rewrite the sdp like you said, so it will contain 
the ip of the rtpproxy (or mediaproxy).  It can be used to proxy the rtp 
traffic to a specific destination.


But the rtpproxy does not necessarily have to be the same machine as 
your Opensips. The rtpproxy is a different tool which will be enabled by 
setting the relevant modparam settings and also calling the rtpproxy in 
your routing-script.


http://voiprookie.blogspot.com/2009/04/rtpproxy-12x-installation.html

maybe there are also other, better tutorials but this link seems to be a 
good start for setting up rtpproxy with opensips.



BR

Max M.


Am 27.08.2010 15:36, schrieb TeleCube - John:
  

Hi,

I have a setup as follows:

1.2.3.4 - Outbound Route
4.5.6.7 - Opensips proxy
4.5.6.8 - Asterisk server

The outbound route will only allow any traffic, signaling and media, via
the 4.5.6.7 IP address. It cannot currently accept media from a
different ip than the signaling.

Currently the astx server sends media direct to the outbound route.

Signaling is fine but I need to force the media stream from the asterisk
server through the opensips proxy.

Am I correct in expecting that nathelper/rtpproxy can do this?

Is it simply a matter of re-writing the sdp detail that will tell the
asterisk server to route media through the opensips ip address?

Any examples will be greatly appreciated.

Thanks,
John

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Re: [OpenSIPS-Users] opensips long shutdown time

[Dave Singer]

Thanks for the reply.

I would also lean to the DB as the problem but I have also experienced it
with about the same frequency when not using any database modules and low or
now calls.
So if you could point me to the patch you refereed to I would appreciate it.
I'll then run it through a scripted process of restarting to try to
reproduce and catch the problem (as time permits).

Thanks again for all your work with development and support!!

Dave (cando)

On Fri, Aug 27, 2010 at 2:02 AM, Bogdan-Andrei Iancu  wrote:

> Hi Dave,
>
> At sthutdown, opensips modules are mainly doing memory cleanup (dialog,
> transactions, caches, etc) or data flush to DB (dialogs, usrloc, etc).
>
> As memory cleanup is predictable as time and not resource consuming, I
> would say it is something related to DB flush - some DB ops that take
> random number of (mili) seconds, depending on the DB server load.
>
> A simple patch to report the shutdown duration of each module can be
> made if you consider it helpful - but I suspect the DB ops.
>
> Regards,
> Bogdan
>
> Dave Singer wrote:
> > Sometimes when I'm do a restart on opensips (using init.d script, with
> > some customization to handle this problem) opensips takes quite a bit
> > of time, like 30 - 50 seconds to stop. Other times it is very quick.
> > I'm using 1.6.2 and 1.6.3.
> > The servers are fairly busy, less then 200 calls per sec, with around
> > 1000 sustained open calls.
> > I am using dialog module and db mode is 3 (on shutdown) with
> > postgresql 8.4. However since sometimes it is very quick with the same
> > call volume it doesn't "feel" like this is the problem. Further I have
> > had it be slow to stop when testing a config that is not using dialog
> > at all and there are no active calls or any sip activity when I try to
> > stop it.
> > My hunch is slightly toward it is waiting on a transaction but not so
> > sure.
> >
> > The pertinent part of my init.d script customizations are for after
> > sending opensips the kill  signal to every 0.1 seconds check if
> > it has stopped yet then start it back up.
> >
> > Needless to say, having it take 50 seconds to restart, and not
> > responding to sip traffic does not make customers happy when your
> > dealing with thousands of call. :(
> >
> > Any thoughts or things to try would be appreciated.
> >
> > Thanks
> >
> > cando
> > 
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS Bootcamp
> 20 - 24 September 2010, Frankfurt, Germany
> www.voice-system.ro
>
>
>
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Re: [OpenSIPS-Users] Disable inactive resource when redirecting

[futurewizard]

Hi Bogdan

How did I disabled the destination ..I simply killed the application.

I have 4 application instances running as 5800, 5801, 5802 and 5803. I
killed of of the application (Which I think is the case I want to test).

Let me write what I'm trying to do?

Use OpenSips as Redirector to these 4 applications. And if any of the
application goes down, I want OpenSips to disable that destination.

Why am I taking this route?

I have application registering as sip:a...@internal_ip:port, however, I will
get call as sip:12...@external_ip or sip:67...@external_ip.

Any thought/help?

Thanks for your time.
-- 
View this message in context: 
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Re: [OpenSIPS-Users] Opensips 1.6.2 to 1.6.3 SST module question

[k1028]

Thank you very much for your response. I will look into the re-invite problem
now.

There is no BYE in the SIPTrace from the SIPTrace module associated to this.
The first Invite is received at 17:15:19 and the last ACK is at 17:16:05
from SIPTrace module. The dialog module send BYE at 17:19:05
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Re: [OpenSIPS-Users] Log authentication errors

[Kennard_White]

Hi Joan,

Interesting idea. There are a few edge-cases you might want to consider:

1. A legitimate client will have periodic auth failure due to expired nonce
serialization (see "nonce_expire" parameter).  This is expected behavior
when the nonce cached by a client is expired (by default every 30sec). The
return code -3 from www_authorize() indicates this case, and the reply to
client has stale=1 parameter.

2. Similar to above, a legitimate client will have auth failure due to
nonce serialization (see "disable_nonce_check" parameter). This also has a
-3 return code; however, the response back to the client does NOT have the
stale=1 parameter.

3. Generally the "first" request doesn't have any credentials at all
(because client doesn't have a nonce), and probably isn't an "attack".
Someone else one else suggested searching for "Authorization" header field
to detect this. You can also detect this via the -4 return code.

Thus, I'm suggesting something like:
   $var(auth_code) = www_authorize(...);
   if ( $var(auth_code) == -1 || $var(auth_code) == -2 ) {
xlog("L_ERR","Auth error for $...@$fd from $si cause $var
(auth_code)");
   }
   if ( $var(auth_code) < 0 ) {
www_challenge(...);
exit;
   }


Also, I wonder if it is possible store the data using db_flatstore. Is
there an interface to to call db_flatstore's insert method from script
land? avpops doesn't seem to have an avp_db_insert() method.

Regards,
Kennard



From:   Joan 
To: OpenSIPS users mailling list 
Date:   08/27/2010 08:25 AM
Subject:[OpenSIPS-Users] Log authentication errors
Sent by:users-boun...@lists.opensips.org



Hello,

Since some time ago there are plenty of hackers trying to stole
accounts and bruteforcing passwords. I would like to log all the
wrong authentications so I can use fail2ban to block those ips.
I've been reading all the mailing list history and I don't see
anything related althought is quite interesting.
I've been trying something and at the moment the best I could do is this:

if (is_method("REGISTER")) {
# authenticate the REGISTER requests
if (!www_authorize("", "subscriber")) {
xlog("L_ERR","Auth error for $...@$fd from $si");
www_challenge("", "0");
exit;
}


So when the device tries to register with a wrong password/username i
will log  the errors. After that I still have to do the fail2ban
script, but that's quite easy.
I'd like to know your opinion about this.

Thanks

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Re: [OpenSIPS-Users] Log authentication errors

[Brett Nemeroff]

On Fri, Aug 27, 2010 at 10:24 AM, Joan  wrote:

So when the device tries to register with a wrong password/username i
> will log  the errors. After that I still have to do the fail2ban
> script, but that's quite easy.
> I'd like to know your opinion about this.
>
>
Joan,
I recommend to all of my clients to use Fail2Ban. I think it's a great tool.
I think you should share your log entry filter and action on the wiki for
others to see. :)
-Brett
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Re: [OpenSIPS-Users] LB under a 64bit Debian Lenny

[DM]

Bogdan,

It works like a charm, thank you!

Kind regards,

Dimitri


2010/8/27 Bogdan-Andrei Iancu 

> yes, update to latest version from branch 1.6
>
> Regards,
> Bogdan
>
> DM wrote:
> > Hi Bogdan,
> >
> > You mean this one?
> >
> > "
> >
> > SVN download of latest stable release (1.6):
> >
> >|# svn co
> https://opensips.svn.sourceforge.net/svnroot/opensips/branches/1.6opensips_1_6|
> > "
> >
> >
> > or do I need to fetch the devel version?
> >
> > Thanks,
> > Dimitri
> >
> > 2010/8/27 Bogdan-Andrei Iancu  > >
> >
> > Dmitri,
> >
> > if you update from SVN, the bug should be fixed. Thanks for reporting
> > and I'm waiting for your confirmation on the fix.
> >
> > Regards,
> > Bogdan
> >
> > Bogdan-Andrei Iancu wrote:
> > > Hi Dimitri,
> > >
> > > I was able to reproduce it - working on a fix now.
> > >
> > > Regards,
> > > Bogdan
> > >
> > > DM wrote:
> > >
> > >> Hi Bogdan,
> > >>
> > >> Any luck regarding this problem?
> > >> No need to hurry, I'm just asking :-)
> > >>
> > >> Dimitri
> > >>
> > >> 2010/8/10 Bogdan-Andrei Iancu  > 
> > >> >>
> > >>
> > >> Hi Dimitri,
> > >>
> > >> Hard to tellI will try to do a setup on 64b machine
> > with more than
> > >> 32.
> > >>
> > >> Let me come back to you...
> > >>
> > >> Regards,
> > >> Bogdan
> > >>
> > >> Dmitri G. wrote:
> > >> > Hi,
> > >> >
> > >> > I have compiled Opensips 1.6.2 (downloaded from opensips
> > >> website) and
> > >> > I have used the LB opensips.cfg file on opensips.org
> > 
> > >> 
> > >> > .
> > >> >
> > >> > I have also added 80 destinations to the load_balancer
> table.
> > >> > For some reason, destinations above Id 30 aren't
> > considered, always
> > >> > gettibg back 500 Service full.
> > >> >
> > >> > In the same box, I have used this source and built a
> > 32bit Opensips
> > >> > with pbuilder, it works fine for all destinations.
> > >> >
> > >> > With using Opensips 1.6.3 (downloaded from opensips.org
> > 
> > >> 
> > >> > ), the errors are the same (with
> 64bit,
> > >> > destinations above ID 30 doesn1t work at all).
> > >> >
> > >> > when I run lb_list, all destinations are listed, and
> > marked as
> > >> active
> > >> > (true for both opensips 1.6.2 and 1.6.3, both 32bit and
> > 64bit)
> > >> >
> > >> > In syslog, there is no errors.
> > >> >
> > >> > What can cause this?
> > >> >
> > >> > Thanks,
> > >> >
> > >> > Dimitri
> > >> >
> > >>
> >
> 
> > >> >
> > >> > ___
> > >> > Users mailing list
> > >> > Users@lists.opensips.org
> >   > >
> > >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > >> >
> > >>
> > >>
> > >> --
> > >> Bogdan-Andrei Iancu
> > >> OpenSIPS Bootcamp
> > >> 20 - 24 September 2010, Frankfurt, Germany
> > >> www.voice-system.ro 
> > 
> > >>
> > >>
> > >> ___
> > >> Users mailing list
> > >> Users@lists.opensips.org 
> > >
> > >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > >>
> > >>
> > >>
> >
> 
> > >>
> > >> ___
> > >> Users mailing list
> > >> Users@lists.opensips.org 
> > >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > >>
> > >>
> > >
> > >
> > >
> >
> >
> > --
> > Bogdan-Andrei Iancu
> > OpenSIPS Bootcamp
> > 20 - 24 September 2010, Frankfurt, Germany
> > www.voice-system.ro 
> >
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org 
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
> > -

[OpenSIPS-Users] pua_bla and naprt/srv failover issue

[Zahid Mehmood]

Hi,
  With pua_bla's default_domain set to point to use a naptr record (srv 
records resolve to two hosts), opensips picks one hostname/IP and only attempts 
to send sip messages to it.  If that host is unavailable, OpenSIPS does not 
attempt to contact the second host returned by the SRV records.  

Is this the correct / expected behavior?  I expected the server to failover and 
try the next entry.

Thanks.

-- 
Zahid


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Re: [OpenSIPS-Users] Log authentication errors

[Bogdan-Andrei Iancu]

Note that by checking the return code of www_authorize() you can get the 
reason of the failure:
http://www.opensips.org/html/docs/modules/1.6.x/auth_db.html#id228268

Regards,
Bogdan

Stanisław Pitucha wrote:
> On 27/08/10 16:36, Joan wrote:
>   
>> At the moment I still have some doubts on where to put the logging
>> part, to minimize the false positives (setting like in the example it
>> marks the first packet as wrong)
>> 
>
> Not tested at all - but something like that should work.
>
> if (is_method("REGISTER")) {
> # authenticate the REGISTER requests
> if (!www_authorize("", "subscriber")) {
> if (is_present_hf("Authorization"))
> xlog("L_ERR","Auth error for $...@$fd from $si");
> www_challenge("", "0");
> exit;
> }
>
> Regards,
> Stan
>
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>
>   


-- 
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
20 - 24 September 2010, Frankfurt, Germany
www.voice-system.ro


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Re: [OpenSIPS-Users] LB under a 64bit Debian Lenny

[Bogdan-Andrei Iancu]

yes, update to latest version from branch 1.6

Regards,
Bogdan

DM wrote:
> Hi Bogdan,
>
> You mean this one?
>
> "
>
> SVN download of latest stable release (1.6):
>
>|# svn co 
> https://opensips.svn.sourceforge.net/svnroot/opensips/branches/1.6 
> opensips_1_6|
> "
>
>
> or do I need to fetch the devel version?
>
> Thanks,
> Dimitri
>   
> 2010/8/27 Bogdan-Andrei Iancu  >
>
> Dmitri,
>
> if you update from SVN, the bug should be fixed. Thanks for reporting
> and I'm waiting for your confirmation on the fix.
>
> Regards,
> Bogdan
>
> Bogdan-Andrei Iancu wrote:
> > Hi Dimitri,
> >
> > I was able to reproduce it - working on a fix now.
> >
> > Regards,
> > Bogdan
> >
> > DM wrote:
> >
> >> Hi Bogdan,
> >>
> >> Any luck regarding this problem?
> >> No need to hurry, I'm just asking :-)
> >>
> >> Dimitri
> >>
> >> 2010/8/10 Bogdan-Andrei Iancu  
> >> >>
> >>
> >> Hi Dimitri,
> >>
> >> Hard to tellI will try to do a setup on 64b machine
> with more than
> >> 32.
> >>
> >> Let me come back to you...
> >>
> >> Regards,
> >> Bogdan
> >>
> >> Dmitri G. wrote:
> >> > Hi,
> >> >
> >> > I have compiled Opensips 1.6.2 (downloaded from opensips
> >> website) and
> >> > I have used the LB opensips.cfg file on opensips.org
> 
> >> 
> >> > .
> >> >
> >> > I have also added 80 destinations to the load_balancer table.
> >> > For some reason, destinations above Id 30 aren't
> considered, always
> >> > gettibg back 500 Service full.
> >> >
> >> > In the same box, I have used this source and built a
> 32bit Opensips
> >> > with pbuilder, it works fine for all destinations.
> >> >
> >> > With using Opensips 1.6.3 (downloaded from opensips.org
> 
> >> 
> >> > ), the errors are the same (with 64bit,
> >> > destinations above ID 30 doesn1t work at all).
> >> >
> >> > when I run lb_list, all destinations are listed, and
> marked as
> >> active
> >> > (true for both opensips 1.6.2 and 1.6.3, both 32bit and
> 64bit)
> >> >
> >> > In syslog, there is no errors.
> >> >
> >> > What can cause this?
> >> >
> >> > Thanks,
> >> >
> >> > Dimitri
> >> >
> >>
> 
> >> >
> >> > ___
> >> > Users mailing list
> >> > Users@lists.opensips.org
>   >
> >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >> >
> >>
> >>
> >> --
> >> Bogdan-Andrei Iancu
> >> OpenSIPS Bootcamp
> >> 20 - 24 September 2010, Frankfurt, Germany
> >> www.voice-system.ro 
> 
> >>
> >>
> >> ___
> >> Users mailing list
> >> Users@lists.opensips.org 
> >
> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>
> >>
> >>
> 
> >>
> >> ___
> >> Users mailing list
> >> Users@lists.opensips.org 
> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>
> >>
> >
> >
> >
>
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS Bootcamp
> 20 - 24 September 2010, Frankfurt, Germany
> www.voice-system.ro 
>
>
> ___
> Users mailing list
> Users@lists.opensips.org 
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


-- 
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OpenSIPS Bootcamp
20 - 24 September 2010, Frankfurt, Germany
www.voice-system.ro


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ht

Re: [OpenSIPS-Users] Route timeout

[Brett Nemeroff]

On Fri, Aug 27, 2010 at 8:33 AM, Wesley Volcov wrote:

> Follow my tests:
> The INVITE was sent by opensiups  to route at 09:03:42
> The 100 TRY  came from route to opensips at 09:03:42
> The 183 came from route to opensips at 09:03:43
> But, at 09:03:47 the was sent to Failure Route by Opensips.
>
> The call is going to Failure Route 5 seconds after 100 TRY and not after
> 183/180.


Wesley,
I had a similar situation as you. Basically I needed a PDD timeout to roll
over to another carrier. The timer here is a timer to a final reply. In
other words to an answer or reject. So what you need to do is in the onreply
route check for a 18X type reply and if you get it, then reset the timer to
the ring timeout, else the timer will still be at 5 seconds. The logic
should be something like:
Attempt call out to carrier
If 5 seconds without a 180/183, goto failure route
if 180/183 receieved onreply, then set timer to 120 seconds

Good Luck!
-Brett
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[OpenSIPS-Users] Force media through the Opensips server

[TeleCube - John]

Hi,

I have a setup as follows:

1.2.3.4 - Outbound Route
4.5.6.7 - Opensips proxy
4.5.6.8 - Asterisk server

The outbound route will only allow any traffic, signaling and media, via 
the 4.5.6.7 IP address. It cannot currently accept media from a 
different ip than the signaling.

Currently the astx server sends media direct to the outbound route.

Signaling is fine but I need to force the media stream from the asterisk 
server through the opensips proxy.

Am I correct in expecting that nathelper/rtpproxy can do this?

Is it simply a matter of re-writing the sdp detail that will tell the 
asterisk server to route media through the opensips ip address?

Any examples will be greatly appreciated.

Thanks,
John

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Re: [OpenSIPS-Users] LB under a 64bit Debian Lenny

[Bogdan-Andrei Iancu]

Hi Dimitri,

I was able to reproduce it - working on a fix now.

Regards,
Bogdan

DM wrote:
> Hi Bogdan,
>
> Any luck regarding this problem?
> No need to hurry, I'm just asking :-)
>
> Dimitri
>
> 2010/8/10 Bogdan-Andrei Iancu  >
>
> Hi Dimitri,
>
> Hard to tellI will try to do a setup on 64b machine with more than
> 32.
>
> Let me come back to you...
>
> Regards,
> Bogdan
>
> Dmitri G. wrote:
> > Hi,
> >
> > I have compiled Opensips 1.6.2 (downloaded from opensips
> website) and
> > I have used the LB opensips.cfg file on opensips.org
> 
> > .
> >
> > I have also added 80 destinations to the load_balancer table.
> > For some reason, destinations above Id 30 aren't considered, always
> > gettibg back 500 Service full.
> >
> > In the same box, I have used this source and built a 32bit Opensips
> > with pbuilder, it works fine for all destinations.
> >
> > With using Opensips 1.6.3 (downloaded from opensips.org
> 
> > ), the errors are the same (with 64bit,
> > destinations above ID 30 doesn1t work at all).
> >
> > when I run lb_list, all destinations are listed, and marked as
> active
> > (true for both opensips 1.6.2 and 1.6.3, both 32bit and 64bit)
> >
> > In syslog, there is no errors.
> >
> > What can cause this?
> >
> > Thanks,
> >
> > Dimitri
> >
> 
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org 
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS Bootcamp
> 20 - 24 September 2010, Frankfurt, Germany
> www.voice-system.ro 
>
>
> ___
> Users mailing list
> Users@lists.opensips.org 
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> 
>
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


-- 
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OpenSIPS Bootcamp
20 - 24 September 2010, Frankfurt, Germany
www.voice-system.ro


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Re: [OpenSIPS-Users] Route timeout

[Wesley Volcov]

Hey Bogdan,

Thank you for replying!!

I've been configured:
modparam("tm", "restart_fr_on_each_reply", 1)
modparam("tm", "fr_timer", 2)
modparam("tm", "fr_inv_timer", 5)

Follow my tests:
The INVITE was sent by opensiups  to route at 09:03:42
The 100 TRY  came from route to opensips at 09:03:42
The 183 came from route to opensips at 09:03:43
But, at 09:03:47 the was sent to Failure Route by Opensips.

The call is going to Failure Route 5 seconds after 100 TRY and not after
183/180.

Am I missing anything?

Regards,

On 27 August 2010 05:27, Bogdan-Andrei Iancu  wrote:

> Hi Wesley,
>
> you can use the restart_fr_on_each_reply param (
> http://www.opensips.org/html/docs/modules/1.6.x/tm.html#id271074)  and set
> the fr_inv_timer to 2 secs.
>
> Regards,
> bogdan
>
> Wesley Volcov wrote:
>
>> Hello All,
>>
>> I've configured the params as fallow:
>> modparam("tm", "fr_timer", 2)
>> modparam("tm", "fr_inv_timer", 10)
>>
>> But, after some tests I'm seeing that, when the destination gateway reply
>> a 100 TRYING, the first timeout(fr_timer) is 'disabled', and the call will
>> enter in failure route just if there is no 200 OK in the time configured in
>> fr_inv_time.
>> What I need, is if the time between de 100 TRYING and 180 RING is more
>> than 2 seconds, the call will enter in failure route.
>>
>> How can I do that?
>>
>> Regards,
>>
>>
>> On 12 August 2010 23:33, Pasan Meemaduma > pasan...@ymail.com>> wrote:
>>
>>Hi Wesley,
>>
>>Hope this is what you seeking.
>>
>>modparam("tm", "fr_inv_timer_avp", "$avp(s:timeout)")
>>
>>and you can load different timeout to avp using like below, this
>>is per username
>>
>>avp_db_load("$ru/username","$avp(s:timeout)");
>>
>>and you can trigger failure route as follow,
>>
>> t_on_failure("1");
>>
>>you can use this in your route to trigger next failure route.
>>
>>
>>failure_route[1]
>>{
>>#alternate route
>>}
>>
>>
>>  
>>*From:* Wesley Volcov >>
>>
>>*To:* OpenSIPS users mailling list >>
>>
>>*Sent:* Friday, August 13, 2010 1:17:43
>>*Subject:* [OpenSIPS-Users] Route timeout
>>
>>Hello All,
>>
>>Is there any way to configure a timeout per route?
>>What I mean is: If my first route doesn't send a reply(100 Trying
>>or any other) in some seconds, the opensips sends the the call to
>>failure route, and then I use failure route to try another route.
>>How can I do that?
>>
>>Regards,
>>
>>-- Wesley Volcov
>>Email: wesleyvol...@gmail.com 
>>Messenger: vol...@live.com 
>>
>>Mobile: +55 11 9989-5348
>>Website: http://volcov.blogspot.com
>>
>>
>>
>>
>> --
>> Wesley Volcov
>> Email: wesleyvol...@gmail.com 
>> Messenger: vol...@live.com 
>>
>> Mobile: +55 11 9989-5348
>> Website: http://volcov.blogspot.com
>> 
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS Bootcamp
> 20 - 24 September 2010, Frankfurt, Germany
> www.voice-system.ro
>
>


-- 
Wesley Volcov
Email: wesleyvol...@gmail.com
Messenger: vol...@live.com
Mobile: +55 11 9989-5348
Website: http://volcov.blogspot.com
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Re: [OpenSIPS-Users] Questions about distributing OpenSIPS

[Jim Dalton]

> >   Is OpenSIPS a registered name?
> >
> yes, it is registered.
> >
> >   If so, how do we reference the registration?
> >
> What you mean ?

Here is an example.  Our documentation will state "OSPrey is a registered
trademark of TransNexus, Inc."  We need a similar statement about OpenSIPS,
but we do not know who owns the OpenSIPS name.

Thank you

Jim D.

> -Original Message-
> From: Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro]
> Sent: Friday, August 27, 2010 3:53 AM
> To: jim.dal...@transnexus.com; OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] Questions about distributing OpenSIPS
> 
> Hi Jim,
> 
> Jim Dalton wrote:
> >
> > A lot of our customers use OpenSIPS.  As a convenience for them, we
> > want to distribute OpenSIPS V1.6.3 in binary form and as source code
> > with our free OSPrey routing server.  What notices and disclaimers
> > about OpenSIPS do we need to include in the distribution?
> >
> For sources, there is the GPL license included. For binaries I guess
> there is nothing extra.
> >
> >   Is OpenSIPS a registered name?
> >
> yes, it is registered.
> >
> >   If so, how do we reference the registration?
> >
> What you mean ?
> 
> Best regards,
> Bogdan
> >
> >
> >
> > Thank you,
> >
> >
> >
> > Jim Dalton
> >
> > www.TransNexus.com 
> >
> >
> >
> > -
> ---
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> 
> 
> --
> Bogdan-Andrei Iancu
> OpenSIPS Bootcamp
> 20 - 24 September 2010, Frankfurt, Germany
> www.voice-system.ro
> 




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Re: [OpenSIPS-Users] Registrar, Proxy and Load-balancer

[Bogdan-Andrei Iancu]

Hi Hema,

yes you can configure OpenSIPS in the same time to act as register, 
proxy and LB - it is just a matter of scripting.

Regards,
Bogdan

hemaram wrote:
> Hi Bogdan, 
>
> We have an implementation where we want the Opensips server to work as a
> Proxy, Registrar and Load-balancer (all three at a time) and the Freeswitch
> cluster will just take care of the PBX functionalities. 
>
> Can this be made possible? From all the wiki pages and mailing lists that I
> read, what I understand is that the Opensips can act either as Proxy and
> Registar or as a Load-balancer at a point of time. 
>
> Looking forward to you reply. Thanks in advance! 
>
> Regards, 
> Hema.
>   


-- 
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OpenSIPS Bootcamp
20 - 24 September 2010, Frankfurt, Germany
www.voice-system.ro


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Re: [OpenSIPS-Users] ERROR Column 'to_tag' cannot be null when using

[Bogdan-Andrei Iancu]

Hi Fernando,

The error comes from the dialog module, not from b2buaWhat is the 
exact combination you are using between of these two ?

Regards,
Bogdan

Fernando Testa wrote:
> Hi all, 
>
> When using opensips-1.6.3-tls and module B2BUA I'm getting the errors:
> Aug 24 00:04:47 vm1 /usr/local/sbin/opensips[11500]: 
> INFO:b2b_logic:b2bl_add_client_list: add [191.0] 
> Aug 24 00:04:47 vm1 /usr/local/sbin/opensips[11500]: 
> INFO:b2b_logic:b2bl_print_clients_list: [0x2b887f9f3810] 
> B2B.191.7216225-> 
> Aug 24 00:04:47 vm1 /usr/local/sbin/opensips[11500]: 
> INFO:b2b_logic:b2bl_print_clients_list: 0 
> Aug 24 00:04:47 vm1 /usr/local/sbin/opensips[11500]: ROUTE B2B 
> INITIATE: is_from_ms: [1] : INVITE 
> ruri:[sip:08001...@192.168.0.41:5070 
> ] from:[sip:cli...@192.168.0.41 
> ] to:[sip:08001...@192.168.0.41 
> ] sourceip:[192.168.0.41] 
> callid:[f1489956-5298-4de1-b4cd-644c3ecb7cea] 
> Aug 24 00:04:47 vm1 /usr/local/sbin/opensips[11502]: B2B ON REPLY 
> ROUTE: Started INVITE rs:[180] from:[sip:cli...@192.168.0.41 
> ] to:[sip:08001...@192.168.0.41 
> ] sourceip:[192.168.0.41] 
> callid:[B2B.191.7216225] 
> Aug 24 00:04:47 vm1 /usr/local/sbin/opensips[11498]: 
> INFO:b2b_logic:b2b_add_dlginfo: Dialog pair: 
> [f1489956-5298-4de1-b4cd-644c3ecb7cea] - [B2B.191.7216225] 
> Aug 24 00:04:47 vm1 /usr/local/sbin/opensips[11498]: 
> ERROR:db_mysql:db_mysql_do_prepared_query: driver error: Column 
> 'to_tag' cannot be null 
> Aug 24 00:04:47 vm1 /usr/local/sbin/opensips[11498]: 
> ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db 
> Aug 24 00:04:47 vm1 /usr/local/sbin/opensips[11498]: B2B ON REPLY 
> ROUTE: Started INVITE rs:[200] from:[sip:cli...@192.168.0.41 
> ] to:[sip:08001...@192.168.0.41 
> ] sourceip:[192.168.0.41] 
> callid:[B2B.191.7216225] 
> (full log available 
> at 
> http://gist.github.com/raw/550192/47544f0ad3d72f42f92337dfa8b1625db86fea1d/opensips.log
>  )
> It seems that B2B is not being able to insert a new entry on dialog 
> table. I'm using mysql. 
>
> -- 
> Fernando Gregianin Testa
> Voice Technology Ltda
> +55 11 35882166
>
> 
>
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OpenSIPS Bootcamp
20 - 24 September 2010, Frankfurt, Germany
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Re: [OpenSIPS-Users] Opensips ACK routing problem

[Bogdan-Andrei Iancu]

Hi Nauman,

The problem is with the RURI in ACK ( the identification of the UAC) - 
it has no proto indication and the default is UDP.

The RURI should contain "transport=tls" param or SIPS schema.

Regards,
Bogdan

Nauman Sulaiman wrote:
> Hi, using Opensips 1.6.2. We are using Opensips as outbound proxy using TLS 
> just for final hop between UAC and Opensips. Other legs of the call will be 
> udp. We have a test set up with Sipgate (but same occurs with other 
> providers) where an incoming INVITE to the UAC via opensips results in 200ok  
> generated by UAC and then the ACK coming back from sipgate server is routed 
> over udp connection as opposed to tls, opensips seems to be ignoring the 
> Route header in the ACK
>
> If the contact header has a sips uri opensips will route it correctly but 
> some voip providers do not like sips uri in contact header and we are only 
> using tls for the final hop so are not using a sips uri for contact header.
>
> Do we have our understanding of the spec wrong or should opensips be routing 
> according to the route header or should it use the contact header. 
>
> Opensips ip 172.230.135.190   UAC ip 81.13.94.206
>
> U 217.10.79.23:5060 -> 172.230.135.190:5060
> ACK sip:9082...@81.13.94.206:59053 SIP/2.0.
> Via: SIP/2.0/UDP 217.10.79.23:5060;branch=z9hG4bK59c7.0bbb2877.2.
> Via: SIP/2.0/UDP 172.20.40.4;branch=z9hG4bK59c7.0bbb2877.2.
> Via: SIP/2.0/UDP 
> 217.10.79.23:5060;received=217.10.68.226;branch=z9hG4bK6a423428.
> Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK6a423428;rport=5060.
> Route: 
> ,.
> From: "anonymous" ;tag=as126bf37e.
> To: ;tag=94gc3N42r9X5D.
> Contact: .
> Call-ID: 181c48541a5ad00d04322ccc0752c...@sipgate.co.uk.
> CSeq: 102 ACK.
> Max-Forwards: 67.
> Content-Length: 0.
> X-hint: rr-enforced.
> .
>
>
> U 172.230.135.190:5060 -> 81.13.94.206:59053
> ACK sip:9082...@81.13.94.206:59053 SIP/2.0.
> Via: SIP/2.0/UDP 172.230.135.190;branch=z9hG4bK59c7.6f77cf2.3.
> Via: SIP/2.0/UDP 
> 217.10.79.23:5060;rport=5060;received=217.10.79.23;branch=z9hG4bK59c7.0bbb2877.2.
> Via: SIP/2.0/UDP 172.20.40.4;branch=z9hG4bK59c7.0bbb2877.2.
> Via: SIP/2.0/UDP 
> 217.10.79.23:5060;received=217.10.68.226;branch=z9hG4bK6a423428.
> Via: SIP/2.0/UDP 217.10.66.71:5060;branch=z9hG4bK6a423428;rport=5060.
> From: "anonymous" ;tag=as126bf37e.
> To: ;tag=94gc3N42r9X5D.
> Contact: .
> Call-ID: 181c48541a5ad00d04322ccc0752c...@sipgate.co.uk.
> CSeq: 102 ACK.
> Max-Forwards: 66.
> Content-Length: 0.
> X-hint: rr-enforced.
>  
>
>
>
>   
>
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>   


-- 
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Re: [OpenSIPS-Users] format of the CDR in OpenCP 1.4 using OpenSIPS 1.6.2

[Bogdan-Andrei Iancu]

Hi Morgan,


Morgan Richomme wrote:
> Hi,
>
> I installed and OpenSIPS 1.6.2 and openCP 1.4.
> I configured the cdrviewer part.
> I am puzzled of the format of the CDRs
>
> basically the acc/cdrs, cron and display mechanism in the openSIPS CP is 
> working fine.
> I got all the information stored in the DB, in the CDRS table (coming 
> from acc processing).
> but in the OpenCP by default I have fields I do not have in the acc 
> table (caller, Callee, Leg Type).
>   
yeah, I asked Alex to remove these fields from the display list, as they 
are not by default present in DB

> Do we have to manually define extra accounting fields in acc to get the 
> correct display in the CP?
>   
yes, as there were long discussion about the definition of the caller 
and callee (like is caller the auth user, the FROM user, some trunk ID, 
etc)...so there is no default callee and caller - you have to define 
them (specifically to your setup) via extra accounting.
> If so, shall the tables (acc & cdrs) be modified accordingly?
>   
yes, plus the missed_call table and the mysql procedure.
> by default the cdrs table in 1.6 is limited to the following fields
> - cdr_id
> - call_start_time
> - duration
> - sip_call_id
> - sip_from_tag
> - sip_to_tag
> - created
>
> thus no caller and callee here..or did I miss something?
>   
no, nothing missing - this is the default ACC set provided by OpenSIPS - 
if you need more you need to use extra_accounting in ACC module, to 
extend the acc-related tables and procedures and to add them into CDRviewer.

Regards,
Bogdan
>
> regards
>
> Morgan
>
>
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Re: [OpenSIPS-Users] opensips long shutdown time

[Bogdan-Andrei Iancu]

Hi Dave,

At sthutdown, opensips modules are mainly doing memory cleanup (dialog, 
transactions, caches, etc) or data flush to DB (dialogs, usrloc, etc).

As memory cleanup is predictable as time and not resource consuming, I 
would say it is something related to DB flush - some DB ops that take 
random number of (mili) seconds, depending on the DB server load.

A simple patch to report the shutdown duration of each module can be 
made if you consider it helpful - but I suspect the DB ops.

Regards,
Bogdan

Dave Singer wrote:
> Sometimes when I'm do a restart on opensips (using init.d script, with 
> some customization to handle this problem) opensips takes quite a bit 
> of time, like 30 - 50 seconds to stop. Other times it is very quick. 
> I'm using 1.6.2 and 1.6.3.
> The servers are fairly busy, less then 200 calls per sec, with around 
> 1000 sustained open calls.
> I am using dialog module and db mode is 3 (on shutdown) with 
> postgresql 8.4. However since sometimes it is very quick with the same 
> call volume it doesn't "feel" like this is the problem. Further I have 
> had it be slow to stop when testing a config that is not using dialog 
> at all and there are no active calls or any sip activity when I try to 
> stop it.
> My hunch is slightly toward it is waiting on a transaction but not so 
> sure.
>
> The pertinent part of my init.d script customizations are for after 
> sending opensips the kill  signal to every 0.1 seconds check if 
> it has stopped yet then start it back up.
>
> Needless to say, having it take 50 seconds to restart, and not 
> responding to sip traffic does not make customers happy when your 
> dealing with thousands of call. :(
>
> Any thoughts or things to try would be appreciated.
>
> Thanks
>
> cando
> 
>
> ___
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


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Re: [OpenSIPS-Users] Route timeout

[Bogdan-Andrei Iancu]

Hi Wesley,

you can use the restart_fr_on_each_reply param 
(http://www.opensips.org/html/docs/modules/1.6.x/tm.html#id271074)  and 
set the fr_inv_timer to 2 secs.

Regards,
bogdan

Wesley Volcov wrote:
> Hello All,
>
> I've configured the params as fallow:
> modparam("tm", "fr_timer", 2)
> modparam("tm", "fr_inv_timer", 10)
>
> But, after some tests I'm seeing that, when the destination gateway 
> reply a 100 TRYING, the first timeout(fr_timer) is 'disabled', and the 
> call will enter in failure route just if there is no 200 OK in the 
> time configured in fr_inv_time.
> What I need, is if the time between de 100 TRYING and 180 RING is more 
> than 2 seconds, the call will enter in failure route.
>
> How can I do that?
>
> Regards,
>
>
> On 12 August 2010 23:33, Pasan Meemaduma  > wrote:
>
> Hi Wesley,
>
> Hope this is what you seeking.
>
> modparam("tm", "fr_inv_timer_avp", "$avp(s:timeout)")
>
> and you can load different timeout to avp using like below, this
> is per username
>
> avp_db_load("$ru/username","$avp(s:timeout)");
>
> and you can trigger failure route as follow,
>
>  t_on_failure("1");
>
> you can use this in your route to trigger next failure route.
>
>
> failure_route[1]
> {
> #alternate route
> }
>
> 
> *From:* Wesley Volcov  >
> *To:* OpenSIPS users mailling list  >
> *Sent:* Friday, August 13, 2010 1:17:43
> *Subject:* [OpenSIPS-Users] Route timeout
>
> Hello All,
>
> Is there any way to configure a timeout per route?
> What I mean is: If my first route doesn't send a reply(100 Trying
> or any other) in some seconds, the opensips sends the the call to
> failure route, and then I use failure route to try another route.
> How can I do that?
>
> Regards,
>
> -- 
> Wesley Volcov
> Email: wesleyvol...@gmail.com 
> Messenger: vol...@live.com 
> Mobile: +55 11 9989-5348
> Website: http://volcov.blogspot.com
>
>
>
>
> -- 
> Wesley Volcov
> Email: wesleyvol...@gmail.com 
> Messenger: vol...@live.com 
> Mobile: +55 11 9989-5348
> Website: http://volcov.blogspot.com
> 
>
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Re: [OpenSIPS-Users] Question about source port on t_relay

[Bogdan-Andrei Iancu]

Hi Daniel,

That is impossible from networking point of view - as you can have a 
single socket bind to a port, and you can have a single TCP connection 
on a socket, it is rather impossible to have multiple TCP connection 
from the same port.

Regards,
Bogdan

Daniel Goepp wrote:
> Yes, you are correct.  I understand that the port in the parameter is 
> the destination, I just was including exactly what we do.  And yes, 
> what we are experiencing is just what you say.  We are binding on port 
> 5060 both UDP and TCP to receive traffic.  Is it possible to force an 
> outbound request on TCP to have a source port of 5060, instead of just 
> requesting one from the OS? 
>
> -dg
>
>
> On Wed, Aug 18, 2010 at 7:53 AM, Bogdan-Andrei Iancu 
> mailto:bog...@voice-system.ro>> wrote:
>
> Hi Daniel,
>
> the param of t_relay() is the destination where to send, not the
> source.
> In UDP, the source will be all the time one of the defined interfaces,
> but for TPC/TLS, when firing a new connection, the kernel allocates a
> new random port for that - I guess this is the case for you, right ?
>
> Regards,
> Bogdan
>
> Daniel Goepp wrote:
> > We are using:
> >
> > t_relay("tcp::5060")
> >
> > However this causes the source port to be someone unpredictable.
>  But
> > if we use UDP, it always comes from 5060 (as we would expect).
> >
> > I'm just looking for some comments on this behavior, and if it is
> > expected or a bug.
> >
> > Thanks
> >
> > -dg
> >
> 
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org 
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
>
> --
> Bogdan-Andrei Iancu
> OpenSIPS Bootcamp
> 20 - 24 September 2010, Frankfurt, Germany
> www.voice-system.ro 
>
>
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>
>
> 
>
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Re: [OpenSIPS-Users] opensips long shutdown time

[Bogdan-Andrei Iancu]

Hi Dave,

Your email did go through - if you do not receive any auto-reply , is 
ok. You can also check the mailing list achive 
(http://lists.opensips.org/pipermail/users/).

About getting an answer - you have to understand that this is a free 
project and also the help you get is for free, so there is no guarantee 
how fast someone will answer you or if any will answer you at all.

Patience is a virtue.

Regards,
Bogdan

Dave Singer wrote:
> If this goes through to everyone on the list just ignore the paragraph 
> below and look at the meat of the problem below it in the forward.
>
> This is/was my first post so I'm not sure exactly sure how things 
> work. It has been a few days and I haven't seen it come out among the 
> other emails from the list and I've seen no reply or find it on the 
> list site. So I don't know if it is waiting for approval after my 
> subscription, my subscriptions approval, someone to actually see it 
> and have a thought on it I don't know. I'm patient but just want 
> to make sure I'm not waiting on myself and not realizing it. :P
>
> Thanks
>
> On Fri, Aug 20, 2010 at 11:06 AM, cando wrote:
>
> Sometimes when I'm do a restart on opensips (using init.d script,
> with some customization to handle this problem) opensips takes
> quite a bit of time, like 30 - 50 seconds to stop. Other times it
> is very quick. I'm using 1.6.2 and 1.6.3.
> The servers are fairly busy, less then 200 calls per sec, with
> around 1000 sustained open calls.
> I am using dialog module and db mode is 3 (on shutdown) with
> postgresql 8.4. However since sometimes it is very quick with the
> same call volume it doesn't "feel" like this is the problem.
> Further I have had it be slow to stop when testing a config that
> is not using dialog at all and there are no active calls or any
> sip activity when I try to stop it.
> My hunch is slightly toward it is waiting on a transaction but not
> so sure.
>
> The pertinent part of my init.d script customizations are for
> after sending opensips the kill  signal to every 0.1 seconds
> check if it has stopped yet then start it back up.
>
> Needless to say, having it take 50 seconds to restart, and not
> responding to sip traffic does not make customers happy when your
> dealing with thousands of calls. :(
>
> Any thoughts or things to try would be appreciated.
>
> Thanks
>
> cando 
>
>
> 
>
> ___
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>   


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Re: [OpenSIPS-Users] Multiple response codes being sent

[Brad Bendy]

That did it, wow I feel stupid now, ooops!

(I forgot to hit send a few days and saw this e-mail open still)

Thanks for the help!

On Tue, 2010-08-24 at 18:45 +0300, Bogdan-Andrei Iancu wrote:

> Looks like a mixing of stateless and statefull replies to me . load 
> the "signaling.so" module and replace the sl_send_reply() for 503 with a 
> send_reply.
> 
> Regards,
> Bogdan
> 
> Brad Bendy wrote:
> > Hi Bogdan,
> >
> > In this case the 503 is being sent from a route block via 
> > sl_send_reply, then with a exit() after the sl_send_reply()
> >
> > The same behavior happens in both failure route and the standard route 
> > block.
> >
> > Im 100% sure ive done something wrong in the script :)
> >
> >
> > On Tue, 2010-08-24 at 17:46 +0300, Bogdan-Andrei Iancu wrote:
> >> Hi Brad,
> >>
> >> I guess you are doing something funny in the script like allowing the 
> >> 302 reply to be relaid out, but having the 503 generated by opensips - 
> >> by chance, do you send the 503 in stateless mode ?
> >>
> >> Regards,
> >> Bogdan
> >>
> >> Brad Bendy wrote:
> >> > Hi Bogdan,
> >> >
> >> > Here is a full trace, breakdown is like this
> >> >
> >> > .2 INVITES to .164
> >> > .164 INVITES TO .168
> >> > .168 sends a 302 to .164
> >> > .164 sends .2 a 503 followed by a 302
> >> >
> >> > .2 should never know about the 302 at all, but it's still getting back 
> >> > to the originating proxy.
> >> >
> >> > We are not using get_redirects() to do anything with the 302 - from 
> >> > some Googling and such it appears that might be needed, just not sure 
> >> > how it would be used.
> >> >
> >> > Thanks for looking at this.
> >> >
> >> > 69.xxx.xxx.2:5060 -> 72.xxx.xxx.164:5060
> >> > INVITE sip:6021112...@72.xxx.xxx.164 SIP/2.0.
> >> > Via: SIP/2.0/UDP 69.xxx.xxx.2:5060;branch=z9hG4bK03578afa;rport.
> >> > From: "Test" ;tag=as4f36ab60.
> >> > To: .
> >> > Contact: .
> >> > Call-ID: 7b3d78d644ab1f7d52ced54236154...@69.xxx.xxx.2 
> >> > . 
> >> > 
> >> > CSeq: 102 INVITE.
> >> > User-Agent: None.
> >> > Max-Forwards: 70.
> >> > Remote-Party-ID: "Test" ;privacy=off;screen=no.
> >> > Date: Tue, 24 Aug 2010 12:01:35 GMT.
> >> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> >> > Supported: replaces.
> >> > Content-Type: application/sdp.
> >> > Content-Length: 281.
> >> > .
> >> > v=0.
> >> > o=root 2921 2921 IN IP4 69.xxx.xxx.2.
> >> > s=session.
> >> > c=IN IP4 69.xxx.xxx.2.
> >> > t=0 0.
> >> > m=audio 12570 RTP/AVP 18 0 101.
> >> > a=rtpmap:18 G729/8000.
> >> > a=fmtp:18 annexb=no.
> >> > a=rtpmap:0 PCMU/8000.
> >> > a=rtpmap:101 telephone-event/8000.
> >> > a=fmtp:101 0-16.
> >> > a=silenceSupp:off - - - -.
> >> > a=ptime:20.
> >> > a=sendrecv.
> >> >
> >> >
> >> > U 72.xxx.xxx.164:5060 -> 69.xxx.xxx.2:5060
> >> > SIP/2.0 100 Giving a try.
> >> > Via: SIP/2.0/UDP 69.xxx.xxx.2:5060;branch=z9hG4bK03578afa;rport=5060.
> >> > From: "Test" ;tag=as4f36ab60.
> >> > To: .
> >> > Call-ID: 7b3d78d644ab1f7d52ced54236154...@69.xxx.xxx.2 
> >> > . 
> >> > 
> >> > CSeq: 102 INVITE.
> >> > Server: OpenSIPS (1.6.2-notls (x86_64/freebsd)).
> >> > Content-Length: 0.
> >> > .
> >> >
> >> >
> >> > U 72.xxx.xxx.164:5060 -> 72.xxx.xxx.168:5060
> >> > INVITE sip:6021112...@72.xxx.xxx.168 SIP/2.0.
> >> > Record-Route: 
> >> > .
> >> > Via: SIP/2.0/UDP 72.xxx.xxx.164;branch=z9hG4bK23ef.2faf83c4.0.
> >> > Via: SIP/2.0/UDP 
> >> > 69.xxx.xxx.2:5060;received=69.xxx.xxx.2;branch=z9hG4bK03578afa;rport=5060.
> >> > From: "Test" ;tag=as4f36ab60.
> >> > To: .
> >> > Contact: .
> >> > Call-ID: 7b3d78d644ab1f7d52ced54236154...@69.xxx.xxx.2 
> >> > . 
> >> > 
> >> > CSeq: 102 INVITE.
> >> > User-Agent: None.
> >> > Max-Forwards: 69.
> >> > Remote-Party-ID: "Test" ;privacy=off;screen=no.
> >> > Date: Tue, 24 Aug 2010 12:01:35 GMT.
> >> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
> >> > Supported: replaces.
> >> > Content-Type: application/sdp.
> >> > Content-Length: 281.
> >> > .
> >> > v=0.
> >> > o=root 2921 2921 IN IP4 69.xxx.xxx.2.
> >> > s=session.
> >> > c=IN IP4 69.xxx.xxx.2.
> >> > t=0 0.
> >> > m=audio 12570 RTP/AVP 18 0 101.
> >> > a=rtpmap:18 G729/8000.
> >> > a=fmtp:18 annexb=no.
> >> > a=rtpmap:0 PCMU/8000.
> >> > a=rtpmap:101 telephone-event/8000.
> >> > a=fmtp:101 0-16.
> >> > a=silenceSupp:off - - - -.
> >> > a=ptime:20.
> >> > a=sendrecv.
> >> >
> >> >
> >> > U 72.xxx.xxx.168:5060 -> 72.xxx.xxx.164:5060
> >> > SIP/2.0 100 Giving a try.
> >> > Via: SIP/2.0/UDP 72.xxx.xxx.164;branch=z9hG4bK23ef.2faf83c4.0.
> >> > Via: SIP/2.0/UDP 
> >> > 69.xxx.xxx.2:5060;received=69.xxx.xxx.2;branch=z9hG4bK03578afa;rport=5060.
> >> > From: "Test" ;tag=as4f36ab60.
> >> > To: .
> >> > Call-ID: 7

Re: [OpenSIPS-Users] Need help on realtime integration OpenSIPS with Freeswitch.

[hemaram]

Hi Bogdan,

We are implementing a similar set up like Hung's. 
We want the Opensips to serve as Proxy, Registrar and Load-balancer (all
three at a time) and the Freeswitch cluster will just take care of the PBX
functionalities.

Can this be made possible? From all the wiki pages and mailing lists that I
read, what I understand is that the Opensips can act either as Proxy and
Registar or as a Load-balancer at a point of time.

Looking forward to you reply. Thanks in advance!

Regards,
Hema.
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[OpenSIPS-Users] Registrar, Proxy and Load-balancer

[hemaram]

Hi Bogdan, 

We have an implementation where we want the Opensips server to work as a
Proxy, Registrar and Load-balancer (all three at a time) and the Freeswitch
cluster will just take care of the PBX functionalities. 

Can this be made possible? From all the wiki pages and mailing lists that I
read, what I understand is that the Opensips can act either as Proxy and
Registar or as a Load-balancer at a point of time. 

Looking forward to you reply. Thanks in advance! 

Regards, 
Hema.
-- 
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Re: [OpenSIPS-Users] Loosing DB connection

[Brad Bendy]

Hi,

Yeah, that's what I thought as well - we ran for a while with 2 MySQL
DB's then a flat file and same thing would happen where if the first one
"lost connection" it would only hit the second one and would not try the
first one until a restart.

Let me know if you want me to perform any testing.

Thanks

On Fri, 2010-08-27 at 11:00 +0300, Bogdan-Andrei Iancu wrote:

> Hi Brad,
> 
> strange, as the db_virtual should try to reconnect to primary DB URL 
> from time to time.
> 
> I need to run some tests on this...
> 
> Thanks and regards,
> Bogdan
> 
> Brad Bendy wrote:
> > I should probably add that im using db_virtual with a single DB server 
> > then flat file in failover mode.
> >
> > Ive had this on 1.6.2 and 1.6.1 (using 1.6.2 currently)
> >
> > On Wed, 2010-08-25 at 07:04 -0700, Brad Bendy wrote:
> >> Hi,
> >>
> >> Sometimes OpenSIPS losses connection to our database server and 
> >> starts writing acc entries via flat file. It appears to happen when 
> >> the DB server is under heavy load. It seems like some sort of timeout 
> >> or something.
> >>
> >> Is there a way to have OpenSIPS re connect to the database without 
> >> having to do a restart? We do not lose any data as its still writing 
> >> to flat file but then we have to import from that.
> >>
> >> Thanks!
> >> ___
> >> Users mailing list
> >> Users@lists.opensips.org 
> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >> 
> >
> > 
> >
> > ___
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> >   
> 
> 


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Re: [OpenSIPS-Users] Loosing DB connection

[Bogdan-Andrei Iancu]

Hi Brad,

strange, as the db_virtual should try to reconnect to primary DB URL 
from time to time.

I need to run some tests on this...

Thanks and regards,
Bogdan

Brad Bendy wrote:
> I should probably add that im using db_virtual with a single DB server 
> then flat file in failover mode.
>
> Ive had this on 1.6.2 and 1.6.1 (using 1.6.2 currently)
>
> On Wed, 2010-08-25 at 07:04 -0700, Brad Bendy wrote:
>> Hi,
>>
>> Sometimes OpenSIPS losses connection to our database server and 
>> starts writing acc entries via flat file. It appears to happen when 
>> the DB server is under heavy load. It seems like some sort of timeout 
>> or something.
>>
>> Is there a way to have OpenSIPS re connect to the database without 
>> having to do a restart? We do not lose any data as its still writing 
>> to flat file but then we have to import from that.
>>
>> Thanks!
>> ___
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>> Users@lists.opensips.org 
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> 
>
> 
>
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Re: [OpenSIPS-Users] Disable inactive resource when redirecting

[Bogdan-Andrei Iancu]

Hi,

How do you disable a destination ?

Regards,
Bogdan

futurewizard wrote:
> Ok that was mess ..let me try again
>
> Hi all
>
> I'm using OpenSips in redirect mode.
>
> How can I find if a resource/destination is inactive and I don't return
> inactive resource's info to caller.
>
> Currently I'm using Dispatcher module as 
>
> ds_select_domain("2", "4");
> sl_send_reply("302", "Redirect");
>
> I tried using 
>
> modparam("dispatch", "ds_ping_method", "OPTIONS")
> modparam("dispatch", "ds_ping_from", ""sip:a...@myip")
> modparam("dispatch", "ds_ping interval", 30)
> modparam("dispatch", "ds_probing_mode", 1)
> modparam("dispatch", "ds_probing_threshhold, 2)
> modparam("dispatcher", "options_reply_codes", "481")
>
>
> When I disable a resource, it still round robin to that resource.
>
> note: I'm very new to OpenSips and any help is appreciated.
>
> Thanks
>
>
>   


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Re: [OpenSIPS-Users] Questions about distributing OpenSIPS

[Bogdan-Andrei Iancu]

Hi Jim,

Jim Dalton wrote:
>
> A lot of our customers use OpenSIPS.  As a convenience for them, we 
> want to distribute OpenSIPS V1.6.3 in binary form and as source code 
> with our free OSPrey routing server.  What notices and disclaimers 
> about OpenSIPS do we need to include in the distribution?
>
For sources, there is the GPL license included. For binaries I guess 
there is nothing extra.
>
>   Is OpenSIPS a registered name?
>
yes, it is registered.
>
>   If so, how do we reference the registration?
>
What you mean ?

Best regards,
Bogdan
>
>  
>
> Thank you,
>
>  
>
> Jim Dalton
>
> www.TransNexus.com 
>
>  
>
> 
>
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Re: [OpenSIPS-Users] Frontier | osipsconsole

[Bogdan-Andrei Iancu]

A hint from Google:
http://www.cpan.org/modules/by-module/XML/perl-xml-modules.html

Regards,
Bogdan

David J. wrote:
>   Any idea how to Install Frontier from Cpan?
>
> I am trying to run osipsconsole but get the following messages.
>
> Can't locate Frontier/RPC2.pm in @INC (@INC contains: /etc/perl 
> /usr/local/lib/perl/5.10.0 /usr/local/share/perl/5.10.0 /usr/lib/perl5 
> /usr/share/perl5 /usr/lib/perl/5.10 /usr/share/perl/5.10 
> /usr/local/lib/site_perl .) at /sbin/osipsconsole line 33.
> BEGIN failed--compilation aborted at /sbin/osipsconsole line 33.
>
>
>
>
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Re: [OpenSIPS-Users] Opensips 1.6.2 to 1.6.3 SST module question

[Bogdan-Andrei Iancu]

Hi,

I see that the signalling requires a re-INVITE after 180 secs (3 
minutes) - the SST + DIALOG module will terminate the dialog if there is 
no such re-INVITE.

Unfortunatelly the trace has no timestamps to see if the dialog module 
fires the BYE after 3 minutes.

The re-INVITE must be fires by UAC (see in 200 OK : Session-Expires: 
180;refresher=uac )

Regards,
Bogdan

k1028 wrote:
> The call is established but terminated after some time.
>
> Here is the SIP trace from siptrace module and debug 5 from Opensips. There
> is no BYE in the SIPTrace. Debug 5 from Opensips did show BYE sent to caller
> and to callee from dialog module. 
>
> INVITE sip:1510495x...@74.x.x.x. SIP/2.0
> Record-Route: 
> Via: SIP/2.0/UDP 74.x.x.x.;branch=z9hG4bKd659.0ab95ac3.0
> Via: SIP/2.0/UDP
> 192.168.8.222:5068;received=173.x.x.x;branch=z9hG4bK-d87543-21170d43ec319d0c-1--d87543-;rport=5068
> Max-Forwards: 69
> Contact: 
> To: 
> From: "";tag=5145635c
> Call-ID: OThhNWNjOWNjZWFmZDkwMmFlNTExYmVmNTM1ZTE2NjM.
> CSeq: 2 INVITE
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS
> Content-Type: application/sdp
> User-Agent: Idefisk
> Content-Length: 220
> Session-Expires: 180
>
> v=0
> o=Idefisk_user 6056184806875838134 13270 IN IP4 192.168.8.222
> s=Idefisk_user
> c=IN IP4 74.x.x.x.
> t=0 0
> m=audio 1306 RTP/AVP 97 101
> a=fmtp:101  0-15
> a=rtpmap:97 iLBC/8000
> a=rtpmap:101 telephone-event/8000
>
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 74.x.x.x.;branch=z9hG4bKd659.0ab95ac3.0;received=74.x.x.x.
> Via: SIP/2.0/UDP
> 192.168.8.222:5068;received=173.x.x.x;branch=z9hG4bK-d87543-21170d43ec319d0c-1--d87543-;rport=5068
> Record-Route: 
> From: "";tag=5145635c
> To: 
> Call-ID: OThhNWNjOWNjZWFmZDkwMmFlNTExYmVmNTM1ZTE2NjM.
> CSeq: 2 INVITE
> User-Agent: iWorld
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Contact: 
> Content-Length: 0
>
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP
> 192.168.8.222:5068;received=173.x.x.x;branch=z9hG4bK-d87543-21170d43ec319d0c-1--d87543-;rport=5068
> Record-Route: 
> From: "";tag=5145635c
> To: ;tag=as3154366e
> Call-ID: OThhNWNjOWNjZWFmZDkwMmFlNTExYmVmNTM1ZTE2NjM.
> CSeq: 2 INVITE
> User-Agent: iWorld
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Contact: 
> Content-Length: 0
>
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 74.x.x.x.;branch=z9hG4bKd659.0ab95ac3.0;received=74.x.x.x.
> Via: SIP/2.0/UDP
> 192.168.8.222:5068;received=173.x.x.x;branch=z9hG4bK-d87543-21170d43ec319d0c-1--d87543-;rport=5068
> Record-Route: 
> From: "";tag=5145635c
> To: ;tag=as3154366e
> Call-ID: OThhNWNjOWNjZWFmZDkwMmFlNTExYmVmNTM1ZTE2NjM.
> CSeq: 2 INVITE
> User-Agent: iWorld
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Contact: 
> Content-Length: 0
>
> SIP/2.0 183 Session Progress
> Via: SIP/2.0/UDP
> 192.168.8.222:5068;received=173.x.x.x;branch=z9hG4bK-d87543-21170d43ec319d0c-1--d87543-;rport=5068
> Record-Route: 
> From: "";tag=5145635c
> To: ;tag=as3154366e
> Call-ID: OThhNWNjOWNjZWFmZDkwMmFlNTExYmVmNTM1ZTE2NjM.
> CSeq: 2 INVITE
> User-Agent: iWorld
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Contact: 
> Content-Type: application/sdp
> Content-Length: 259
>
> v=0
> o=root 7723 7723 IN IP4 64.x.x.x
> s=session
> c=IN IP4 74.x.x.x.
> t=0 0
> m=audio 1304 RTP/AVP 97 101
> a=rtpmap:97 iLBC/8000
> a=fmtp:97 mode=30
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:30
> a=sendrecv
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 74.x.x.x.;branch=z9hG4bKd659.0ab95ac3.0;received=74.x.x.x.
> Via: SIP/2.0/UDP
> 192.168.8.222:5068;received=173.x.x.x;branch=z9hG4bK-d87543-21170d43ec319d0c-1--d87543-;rport=5068
> Record-Route: 
> From: "";tag=5145635c
> To: ;tag=as3154366e
> Call-ID: OThhNWNjOWNjZWFmZDkwMmFlNTExYmVmNTM1ZTE2NjM.
> CSeq: 2 INVITE
> User-Agent: iWorld
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Contact: 
> Content-Type: application/sdp
> Content-Length: 259
>
> v=0
> o=root 7723 7724 IN IP4 64.x.x.x
> s=session
> c=IN IP4 64.x.x.x
> t=0 0
> m=audio 50450 RTP/AVP 97 101
> a=rtpmap:97 iLBC/8000
> a=fmtp:97 mode=30
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:30
> a=sendrecv
>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.8.222:5068;received=173.x.x.x;branch=z9hG4bK-d87543-21170d43ec319d0c-1--d87543-;rport=5068
> Record-Route: 
> From: "";tag=5145635c
> To: ;tag=as3154366e
> Call-ID: OThhNWNjOWNjZWFmZDkwMmFlNTExYmVmNTM1ZTE2NjM.
> CSeq: 2 INVITE
> User-Agent: iWorld
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Contact: 
> Content-Type: application/sdp
> Content-Length: 259
> Session-Expires: 180;refresher=uac
>
> v=0
> o=root 7723 7724 IN IP4 64.x.x.x
> s=session
> c=IN IP4 74.x.x.x.
> t=0

Re: [OpenSIPS-Users] opensips process dying (dialplan module?)

[Bogdan-Andrei Iancu]

Perfect!

Regards,
Bogdan

Adelson O. Junior wrote:
> Hey Bogdan,
>
> I changed it and now it is working.
> Thank you very much.
>
> Adelson.
>
> On Wed, Aug 25, 2010 at 12:40 PM, Bogdan-Andrei Iancu
>  wrote:
>   
>> In SVN, this is already changed from attrs_avp  to gw_attrs_avp
>>
>> Regards,
>> Bogdan
>>
>>
>> Adelson O. Junior wrote:
>> 
>>> Hi Bogdan,
>>>
>>> Thanks for replying.
>>>
>>> I've updated for 1.6.3 svn tag  and I'am getting an error about
>>> drouting modparam: "attrs_avp".
>>> Note: this modparam is working with the 1.6.3 package from mirrors.
>>> This is occuring in svn tag only.
>>>
>>> Thanks,
>>> Adelson.
>>>
>>> Aug 24 19:28:24 OPENSIPS02 opensips: ERROR:core:set_mod_param_regex:
>>> parameter  not found in module 
>>> Aug 24 19:28:24 OPENSIPS02 opensips: CRITICAL:core:yyerror: parse
>>> error in config file, line 99, column 36-37: Parameter 
>>>  not found in module  - can't set
>>> Aug 24 19:28:24 OPENSIPS02 opensips: ERROR:core:main: bad config file
>>> (1 errors)
>>>
>>> On Tue, Aug 24, 2010 at 6:07 AM, Bogdan-Andrei Iancu
>>>  wrote:
>>>
>>>   
 Hello Adelson,

 better upgrade to latest 1.6 version (1.6.3) from SVN - it contains all
 recent fixes on 1.6 branch. Take care that the regexp engine for
 dialplan module was changed from 1.6.2 to 1.6.3 as old one (trex) was
 bogus and unmaintained. See
 http://www.opensips.org/Resources/DocsMigration162to163

 The upgrade should be be smooth, no script or DB changes.

 Regards,
 Bogdan


 Adelson O. Junior wrote:

 
> Hi list,
>
> my opensips process is dying constantly.
>
> According the dump core, it seems (for me) to be a function in
> dialplan that is causing it.
> Follows the dump from gdb.
> Is someone facing this kind of problem?
>
> opensips version is: opensips 1.6.1-notls (i386/linux)
>
> Thanks,
>
> 
>
>   
>>>   
> --
> Adelson
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>   
 --
 Bogdan-Andrei Iancu
 OpenSIPS Bootcamp
 20 - 24 September 2010, Frankfurt, Germany
 www.voice-system.ro


 ___
 Users mailing list
 Users@lists.opensips.org
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>>>
>>>
>>>   
>> --
>> Bogdan-Andrei Iancu
>> OpenSIPS Bootcamp
>> 20 - 24 September 2010, Frankfurt, Germany
>> www.voice-system.ro
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>> 
>
>
>
>   


-- 
Bogdan-Andrei Iancu
OpenSIPS Bootcamp
20 - 24 September 2010, Frankfurt, Germany
www.voice-system.ro


___
Users mailing list
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http://lists.opensips.org/cgi-bin/mailman/listinfo/users