Re: [OpenSIPS-Users] opensips - hiding interconnection information

[DanB]

Hi Julien,
For such scenarios you would need a B2BUA ( I personally use B2bua.org
but Yate, Freeswitch,  Asterisk can do it too).
One more concern with carrier traffic would be the SDP body, which can
reveal information about the originator too.

Cheers,
DanB

On Wed, Jul 1, 2009 at 7:08 AM, Julien Chavanton wrote:

>  Hi, I have a business usage security concern.
>
> If we take this scenario :
>
> carrierA --SIP--> carrierB --SIP--> carrierC
>
> CarrierB may not want carrierC to find out about carrierA (IP address)
>
> I beleive "record-routing" and "via header" can provide
> unwanted information,
> any suggestion/options on how we can deal with such requirements using a
> SIP proxy as openSIPS ?
>
> Regards,
> Julien
>
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[OpenSIPS-Users] opensips - hiding interconnection information

[Julien Chavanton]

Hi, I have a business usage security concern.
 
If we take this scenario :
 
carrierA --SIP--> carrierB --SIP--> carrierC
 
CarrierB may not want carrierC to find out about carrierA (IP address)
 
I beleive "record-routing" and "via header" can provide unwanted information,
any suggestion/options on how we can deal with such requirements using a SIP 
proxy as openSIPS ?  
 
Regards,
Julien
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Re: [OpenSIPS-Users] Mediaproxy ver 2.3.4 - Conntrack meets Music on Hold

[Dan Pascu]

On 30 Jun 2009, at 19:11, Ruud Klaver wrote:

>> 3) detect this special case and change existing conntrack to point to
>> the MoH sever, putting it back later
>
> We actually considered implementing this, but not for this particular
> scenario. Some modems seem to switch IP address every X hours. If this
> change is during a relayed call, the RTP from one end will suddenly
> come from a new IP address, like in your scenario. What could be done
> is that the relay keeps listening on the allocated ports in userspace
> for traffic from a different host and switch the conntrack rule to the
> new IP if there is no traffic coming in anymore from the old IP.

This will not work in this case. The case you mention was possible  
because the media relay could detect if the stream coming from the new  
IP:port was actually the same stream by looking to the serial number  
present in the RTP. Since that was the same, it could assume that it's  
the same sender that just happened to have his IP changed during the  
call. In Stuart's case, since it's a new SIP dialog, with a new RTP  
stream sent by a different device it will have a different serial  
number. The only way to work would be to accept any stream which is  
opening Pandora's box for official media hijacking.

--
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Re: [OpenSIPS-Users] Mediaproxy ver 2.3.4 - Conntrack meets Music on Hold

[Dan Pascu]

On 30 Jun 2009, at 15:41, Stuart Marsden wrote:

> Hi,
>
> I am testing a mediaproxy sever using 2.3.4
>
> The tests I have done so far have worked, however there is one  
> scenario
> where once the call has been setup a phone can generate Music on Hold
>
> It does this by making a 2nd call (unrelated to the first at the SIP
> level) and passes what it thinks is  the IP:port of the 1st call far  
> end
> (actually the mediaproxy) to the MoH server.
>
> the MoH sever then sends audio to the mediaproxy.
>
> However since the source of the audio is not the phone, conntrack is  
> not
> forwarding on the audio to the other end.
>
> Options as far as I can see are
>
> 1) remove the restriction on the source IP on who is allowed to  
> forward
> rtp - from what I have ready this probably wont work

That is not an arbitrary restriction. That is how the forwarding works  
at the conntrack level. You have four IP:port pairs (2 a tthe  
endpoints and 2 on the media relay machine). The conntrack rule simply  
forwards the packets between the far endpoints using the 2 local  
IP:port pairs as a meeting ground. You cannot simply mix in data  
coming from some other random IP:port source that was not initially  
specified in the conntrack rule when it was created.

>
> 2) detect this special case and construct a temporary additional  "1
> way"  conntrack just for the MoH
> 3) detect this special case and change existing conntrack to point to
> the MoH sever, putting it back later
>

How exactly do you imagine that this special case can be detected,  
since you said that the second call is unrelated to the first at the  
SIP level? You want to match the proposed IP:port of some stream with  
whatever sessions you already have running and simply assume they are  
related? That is unfeasible and naive. If such a thing would be  
implemented, it will open the doors for hijacking RTP streams with  
official support from the SIP proxy and the media relay.

The proper way to do it (which by the way is already supported by  
mediaproxy) is to send a re-INVITE inside the same dialog with the  
proper IP:port in the stream pointing to the MOH, not to send a  
completely unrelated INVITE that creates a new dialog and hope that  
the proxy and the media relay will somehow figure out that they are  
related and magically bridge them.

--
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[OpenSIPS-Users] nathelper has_sdp?

[Brett Nemeroff]

hello all,I'm using the TLS version of opensips 1.5.1

I can't seem to find an actual source reference where has_sdp shows up. What
module/version is this in? Docs show it in 1.5.

-Brett
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Re: [OpenSIPS-Users] LDAP Authentication

[Gavin Henry]

These are my points too and how I thought the auth should work. But
you need some kind of mapping here for user dns etc. ?

On 30/06/2009, Alan Rubin  wrote:
> Bogdan,
>
> I'm not an LDAP expert either, but I will try to explain the scenario
> better.  As you said, the LDAP bind is static - done once in the
> beginning and sourced from the ldap.cfg file.  Unfortunately, we have a
> filter on our LDAP server that prevents ordinary users from seeing the
> password field in the LDAP entry.  The way we verify authentication in
> our environment is by dynamically substituting the LDAP bind DN with the
> client's uid (and password) and making a simple LDAP query using that
> uid.  If that bind is successful, then we know that the password is
> correct.  It doesn't seem like there is anyway to configure opensips in
> that manner.
>
> The aim, with LDAP, was to have a single-signon environment for our LAN
> and SIP accounts.  This doesn't seem possible, unless you or anyone else
> on the list has any further suggestions.  We could use kerberos/AD
> authentication from the client if that is a possibility.
>
> Regards,
>
>
> Alan Rubin
>
> -Original Message-
> From: Bogdan-Andrei Iancu [mailto:bog...@voice-system.ro]
> Sent: Monday, 29 June 2009 10:13 PM
> To: Alan Rubin
> Cc: users@lists.opensips.org
> Subject: Re: [OpenSIPS-Users] LDAP Authentication
>
> Hi Alan,
>
> I'm not an LDAP expert to get into details about how ldap should be
> configured or soWhat I can tell is that the bind is static (only
> once done at the beginning at that's it)Can you send me a link or
> something to read more about what this dynamic bind means in LDAP ?
>
> Thanks and regards,
> Bogdan
>
> Alan Rubin wrote:
>> Bogdan,
>>
>> Apparently the email administrator had a regex on the SMTP gateway to
>> reject messages with pass (and) word (combined) because of previous
>> users succumbing to phishing exercises.  It may work now, but I will
>> continue to check the archives. Oh well.
>>
>> Regarding:
>> "Now, going to the actual issue, the problem is related to password -
>> about how the client and server (ldap) are keeping the password - do
>> they both keep it same format (like plain text) ?
>>
>> Regards,
>> Bogdan"
>>
>> I think I've figured out the issue, although I don't believe there is
> a
>> solution.  Hopefully you can verify, either way.
>>
>> The bind user in the ldap.cfg file does not have the privilege to
>> retrieve the pass  word field from our LDAP directory.  The only way
> our
>> LDAP setup is supposed to work is by binding using the
>> user-to-be-authenticated directly with the LDAP directory server.  It
> is
>> my understanding, and this is where you can verify or correct me, that
>> opensips and the LDAP module can not change the bind user dynamically.
>>
>> Regards,
>>
>> Alan Rubin
>>
>
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Re: [OpenSIPS-Users] Help needed to change database RW user/password

[Uwe Kastens]

Hi,

what kind of db are you using? Do you know how to change the password on
your database backend? If so, you should use the same password for the
user in your database and in the opensips config.

BR

Uwe



srikanth R schrieb:
> Hi,
>  
> I have got my SIP proxy up and running, but I am able to work only with
> the username 'opensips' and the password 'opensipsrw'. When I change the
> username and password in opensipsctlrc file I get an error message
> saying access denied. I am not asure about which variables to comment
> out and which variables to leave as such. Any help in this regard would
> be appreciated.
>  
> Thanks,
> Srikanth
> 
> 
> 
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> Explorer 8
> .
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[OpenSIPS-Users] Help needed to change database RW user/password

[srikanth R]

Hi,
 
I have got my SIP proxy up and running, but I am able to work only with the 
username 'opensips' and the password 'opensipsrw'. When I change the username 
and password in opensipsctlrc file I get an error message saying access denied. 
I am not asure about which variables to comment out and which variables to 
leave as such. Any help in this regard would be appreciated.
 
Thanks,
Srikanth


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Re: [OpenSIPS-Users] ERROR:core:db_allocate_rows: no memory left - Please help

[Bogdan-Andrei Iancu]

Mani,

Thanks for the feedback - I did the official backport to 1.5 also.

Regards,
Bogdan

mani sivaraman wrote:
> Great. I don't see that error any more after applying the patch.
> Thanks
>
> On Mon, Jun 29, 2009 at 12:01 PM, Bogdan-Andrei Iancu 
> mailto:bog...@voice-system.ro>> wrote:
>
> Hi Mani,
>
> if this will make the testing easier, here is the patch you can
> apply against 1.5.
>
> Regards,
> Bogdan
>
> mani sivaraman wrote:
>
> Thanks Bogdan, I will definitely try the devel (1.6). I cannot
> install 1.6 on the current server I have. I have to setup
> another box and then try it out. I will let you know. Mean
> while I'm trying hard to make xmpp2simple presence status
> work. Any help or pointer would be appreciated.
>
> Thanks
> Mani
>
> On Mon, Jun 29, 2009 at 11:04 AM, Bogdan-Andrei Iancu
> mailto:bog...@voice-system.ro>
>  >> wrote:
>
>Hi Mani,
>
>I fixed this on 1.6 (devel) version by adding fetch support
> in the
>PUA module, so now it can add as many row as possible
> without any
>mem issues. Could you give it a try? if you confirm the fix, I
>will do a backport to 1.5 also.
>
>Best regards,
>Bogdan
>
>mani sivaraman wrote:
>
>I'm using opensips 1.5.1 for the past 1 week. Suddenly
> I get
>this error below when starting. What is going on.
>
>
>
>Jun 26 15:52:14 [2535] ERROR:core:db_allocate_rows: no
> memory left
>Jun 26 15:52:14 [2535]
> ERROR:db_mysql:db_mysql_convert_rows:
>no private memory left
>Jun 26 15:52:14 [2535]
> ERROR:db_mysql:db_mysql_convert_result:
>error while converting rows
>Jun 26 15:52:14 [2535] DBG:core:db_free_columns: freeing
>result columns at 0x81a9b78
>Jun 26 15:52:14 [2535]
> ERROR:db_mysql:db_mysql_store_result:
>error while converting result
>Jun 26 15:52:14 [2535] ERROR:core:db_do_query: error while
>storing resultJun 26 15:52:14 [2535] ERROR:pua:db_restore:
>while querrying table
>Jun 26 15:52:14 [2535] ERROR:pua:mod_init: while restoring
>hash_table
>Jun 26 15:52:14 [2535] ERROR:core:init_mod: failed to
>initialize module pua
>Jun 26 15:52:14 [2535] ERROR:core:main: error while
>initializing modules
>Jun 26 15:52:14 [2535] DBG:xmpp:destroy: cleaning up...
>Jun 26 15:52:14 [2535] DBG:pua:destroy: destroying
> module ...
>Jun 26 15:52:14 [2535] DBG:db_mysql:db_mysql_submit_query:
>discon reset for 135960296
>Jun 26 15:52:14 [2535] DBG:core:pool_remove: removing
>connection from the pool
>Jun 26 15:52:14 [2535] DBG:presence_xml:destroy: start
>Jun 26 15:52:14 [2535] NOTICE:presence:destroy: destroy
> module ...
>Jun 26 15:52:14 [2535] DBG:xlog:destroy: destroy module...
>Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: tm_shutdown
> : start
>Jun 26 15:52:14 [2535] DBG:tm:unlink_timer_lists: emptying
>DELETE list
>Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: emptying
> hash table
>Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: releasing timers
>Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: removing
> semaphores
>Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: destroying
> callback
>lists
>Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: tm_shutdown
> : done
>Jun 26 15:52:14 [2535] DBG:core:destroy_tls: entered
>Jun 26 15:52:14 [2535] DBG:core:shm_mem_destroy:
>Jun 26 15:52:14 [2535] DBG:core:shm_mem_destroy: destroying
>the shared memory lock
>
>  
>  
> 
>
>
>
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Re: [OpenSIPS-Users] mysql database connection error

[Bogdan-Andrei Iancu]

To be honest, this happened to me also couple of times and I'm flirting 
with the idea to set the db_url to empty string in all the 
modulesbetter explicitly set them, rather smashing your head with 
things like this.

Regards,
Bogdan

Gordon Ross wrote:
> Yeah, I think that's a bad idea as well. It would be much better, IMHO, if a 
> DB URI is missing, for OpenSIPS to say that you've been a plonker and 
> forgotten to set one, rather than leading you down a garden path.
>
> GTG
> 
> From: users-boun...@lists.opensips.org [users-boun...@lists.opensips.org] On 
> Behalf Of Brett Nemeroff [br...@nemeroff.com]
> Sent: 30 June 2009 18:50
> To: Uwe Kastens
> Cc: users@lists.opensips.org
> Subject: Re: [OpenSIPS-Users] mysql database connection error
>
> I think the whole idea of a "default db_url" is a mistake... that's my 
> personal opinion. Thanks for helping me find this..
>
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Re: [OpenSIPS-Users] mysql database connection error

[Gordon Ross]

Yeah, I think that's a bad idea as well. It would be much better, IMHO, if a DB 
URI is missing, for OpenSIPS to say that you've been a plonker and forgotten to 
set one, rather than leading you down a garden path.

GTG

From: users-boun...@lists.opensips.org [users-boun...@lists.opensips.org] On 
Behalf Of Brett Nemeroff [br...@nemeroff.com]
Sent: 30 June 2009 18:50
To: Uwe Kastens
Cc: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] mysql database connection error

I think the whole idea of a "default db_url" is a mistake... that's my personal 
opinion. Thanks for helping me find this..

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Re: [OpenSIPS-Users] mysql database connection error

[Uwe Kastens]

Hi Brett,

good to know

BR

Uwe


Brett Nemeroff schrieb:
> Hey.. Ok I found the error and I feel like a complete idiot. I had
> modules loaded without a db_url specified for it (one module's db_url
> was missing).
> 
> I think the whole idea of a "default db_url" is a mistake... that's my
> personal opinion. Thanks for helping me find this..
> -Brett
> 
> 
> On Tue, Jun 30, 2009 at 12:48 PM, Uwe Kastens  > wrote:
> 
> Hi Brett,
> 
> Could be broken libs. I would start opensips with strace and look for
> errors.
> 
> Which OS are you using?
> 
> BR
> 
> Uwe
> 
> Brett Nemeroff schrieb:
> > Still can't connect :( What could I be doing wrong? I wonder if
> it's the
> > mysql client libs somehow?!
> >
> > Access-rights
> > for USER 'opensips', from HOST 'localhost', to DB 'opensips'
> > +-+---+ +-+---+
> > | Select_priv | Y | | Lock_tables_priv | Y |
> > | Insert_priv | Y | | Execute_priv| Y |
> > | Update_priv | Y | | Repl_slave_priv | N |
> > | Delete_priv | Y | | Repl_client_priv | N |
> > | Create_priv | Y | | Create_view_priv | Y |
> > | Drop_priv   | Y | | Show_view_priv  | Y |
> > | Reload_priv | N | | Create_routine_priv | Y |
> > | Shutdown_priv   | N | | Alter_routine_priv | Y |
> > | Process_priv| N | | Create_user_priv | N |
> > | File_priv   | N | | Ssl_type| ? |
> > | Grant_priv  | N | | Ssl_cipher  | ? |
> > | References_priv | Y | | X509_issuer | ? |
> > | Index_priv  | Y | | X509_subject| ? |
> > | Alter_priv  | Y | | Max_questions   | 0 |
> > | Show_db_priv| N | | Max_updates | 0 |
> > | Super_priv  | N | | Max_connections | 0 |
> > | Create_tmp_table_priv | Y |   | Max_user_connections | 0 |
> > +-+---+ +-+---+
> > NOTE:A password is required for user `opensips' :-(
> >
> > The following rules are used:
> >  db:
> >
> 
> '127.0.0.1','opensips','opensips','Y','Y','Y','Y','Y','Y','N','Y','Y','Y','Y','Y','Y','Y','Y','Y','Y'
> >  host  : 'Not processed: host-field is not empty in db-table.'
> >  user  :
> >
> 
> '%','opensips','*4DDB979A2666D0CF0A83FCCED820A64E8EBB6AFD','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','','','','','0','0','0','0'
> >
> >
> >
> > On Tue, Jun 30, 2009 at 11:01 AM, Uwe Kastens  
> > >> wrote:
> >
> > Hmm,
> >
> > And you did change the host in the opensips url from localhost to
> > 127.0.0.1?
> >
> > For a test change the 127.0.0.1 in the mysql to % and try
> again. And
> > mabye change the password with update user set
> > password=password("yourpassword") where user="opensips";
> >
> >
> > Your password entry looks like from a elder version.
> >
> > BR
> >
> > Uwe
> >
> >
> > Brett Nemeroff schrieb:
> > > Here is what I got:
> > > password too long? really? it's not that long.. 
> > >
> > >
> > > mysqlaccess Version 2.06, 20 Dec 2000
> > > By RUG-AIV, by Yves Carlier (yves.carl...@rug.ac.be
> 
> > >
> > >    >>)
> > > Changes by Steve Harvey (s...@vex.net 
> >
> >   >>)
> > > This software comes with ABSOLUTELY NO WARRANTY.
> > > Password for MySQL superuser root:
> > >
> > > Access-rights
> > > for USER 'opensips', from HOST 'localhost', to DB 'opensips'
> > > +-+---+ +-+---+
> > > | Select_priv | Y | | Lock_tables_priv | Y |
> > > | Insert_priv | Y | | Execute_priv| Y |
> > > | Update_priv | Y | | Repl_slave_priv | N |
> > > | Delete_priv | Y | | Repl_client_priv | N |
> > > | Create_priv | Y | | Create_view_priv | Y |
> > > | Drop_priv   | Y | | Show_view_priv  | Y |
> > > | Reload_priv | N | | Create_routine_priv | Y |
> > > | Shutdown_priv   | N | | Alter_routine_priv | Y |
> > > | Pr

Re: [OpenSIPS-Users] mysql database connection error

[Brett Nemeroff]

Hey.. Ok I found the error and I feel like a complete idiot. I had modules
loaded without a db_url specified for it (one module's db_url was missing).
I think the whole idea of a "default db_url" is a mistake... that's my
personal opinion. Thanks for helping me find this..
-Brett


On Tue, Jun 30, 2009 at 12:48 PM, Uwe Kastens  wrote:

> Hi Brett,
>
> Could be broken libs. I would start opensips with strace and look for
> errors.
>
> Which OS are you using?
>
> BR
>
> Uwe
>
> Brett Nemeroff schrieb:
> > Still can't connect :( What could I be doing wrong? I wonder if it's the
> > mysql client libs somehow?!
> >
> > Access-rights
> > for USER 'opensips', from HOST 'localhost', to DB 'opensips'
> > +-+---+ +-+---+
> > | Select_priv | Y | | Lock_tables_priv | Y |
> > | Insert_priv | Y | | Execute_priv| Y |
> > | Update_priv | Y | | Repl_slave_priv | N |
> > | Delete_priv | Y | | Repl_client_priv | N |
> > | Create_priv | Y | | Create_view_priv | Y |
> > | Drop_priv   | Y | | Show_view_priv  | Y |
> > | Reload_priv | N | | Create_routine_priv | Y |
> > | Shutdown_priv   | N | | Alter_routine_priv | Y |
> > | Process_priv| N | | Create_user_priv | N |
> > | File_priv   | N | | Ssl_type| ? |
> > | Grant_priv  | N | | Ssl_cipher  | ? |
> > | References_priv | Y | | X509_issuer | ? |
> > | Index_priv  | Y | | X509_subject| ? |
> > | Alter_priv  | Y | | Max_questions   | 0 |
> > | Show_db_priv| N | | Max_updates | 0 |
> > | Super_priv  | N | | Max_connections | 0 |
> > | Create_tmp_table_priv | Y |   | Max_user_connections | 0 |
> > +-+---+ +-+---+
> > NOTE:A password is required for user `opensips' :-(
> >
> > The following rules are used:
> >  db:
> >
> '127.0.0.1','opensips','opensips','Y','Y','Y','Y','Y','Y','N','Y','Y','Y','Y','Y','Y','Y','Y','Y','Y'
> >  host  : 'Not processed: host-field is not empty in db-table.'
> >  user  :
> >
> '%','opensips','*4DDB979A2666D0CF0A83FCCED820A64E8EBB6AFD','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','','','','','0','0','0','0'
> >
> >
> >
> > On Tue, Jun 30, 2009 at 11:01 AM, Uwe Kastens  > > wrote:
> >
> > Hmm,
> >
> > And you did change the host in the opensips url from localhost to
> > 127.0.0.1?
> >
> > For a test change the 127.0.0.1 in the mysql to % and try again. And
> > mabye change the password with update user set
> > password=password("yourpassword") where user="opensips";
> >
> >
> > Your password entry looks like from a elder version.
> >
> > BR
> >
> > Uwe
> >
> >
> > Brett Nemeroff schrieb:
> > > Here is what I got:
> > > password too long? really? it's not that long.. 
> > >
> > >
> > > mysqlaccess Version 2.06, 20 Dec 2000
> > > By RUG-AIV, by Yves Carlier (yves.carl...@rug.ac.be
> > 
> > > >)
> > > Changes by Steve Harvey (s...@vex.net 
> > >)
> > > This software comes with ABSOLUTELY NO WARRANTY.
> > > Password for MySQL superuser root:
> > >
> > > Access-rights
> > > for USER 'opensips', from HOST 'localhost', to DB 'opensips'
> > > +-+---+ +-+---+
> > > | Select_priv | Y | | Lock_tables_priv | Y |
> > > | Insert_priv | Y | | Execute_priv| Y |
> > > | Update_priv | Y | | Repl_slave_priv | N |
> > > | Delete_priv | Y | | Repl_client_priv | N |
> > > | Create_priv | Y | | Create_view_priv | Y |
> > > | Drop_priv   | Y | | Show_view_priv  | Y |
> > > | Reload_priv | N | | Create_routine_priv | Y |
> > > | Shutdown_priv   | N | | Alter_routine_priv | Y |
> > > | Process_priv| N | | Create_user_priv | N |
> > > | File_priv   | N | | Ssl_type| ? |
> > > | Grant_priv  | N | | Ssl_cipher  | ? |
> > > | References_priv | Y | | X509_issuer | ? |
> > > | Index_priv  | Y | | X509_subject| ? |
> > > | Alter_priv  | Y | | Max_questions   | 0 |
> > > | Show_db_priv| N | | Max_updates | 0 |
> > > | Super_priv  | N | | Max_connections | 0 |
> > > | Create_tmp_table_priv | Y |   | Max_user_connections | 0
> |
> > > +-+---+ +-+---+
> > > NOTE:A password is required for user `opensips' :-(
> > >
> > > The following rules are used:
> > >  db:
> > 

Re: [OpenSIPS-Users] mysql database connection error

[Uwe Kastens]

Hi Brett,

Could be broken libs. I would start opensips with strace and look for
errors.

Which OS are you using?

BR

Uwe

Brett Nemeroff schrieb:
> Still can't connect :( What could I be doing wrong? I wonder if it's the
> mysql client libs somehow?!
> 
> Access-rights
> for USER 'opensips', from HOST 'localhost', to DB 'opensips'
> +-+---+ +-+---+
> | Select_priv | Y | | Lock_tables_priv | Y |
> | Insert_priv | Y | | Execute_priv| Y |
> | Update_priv | Y | | Repl_slave_priv | N |
> | Delete_priv | Y | | Repl_client_priv | N |
> | Create_priv | Y | | Create_view_priv | Y |
> | Drop_priv   | Y | | Show_view_priv  | Y |
> | Reload_priv | N | | Create_routine_priv | Y |
> | Shutdown_priv   | N | | Alter_routine_priv | Y |
> | Process_priv| N | | Create_user_priv | N |
> | File_priv   | N | | Ssl_type| ? |
> | Grant_priv  | N | | Ssl_cipher  | ? |
> | References_priv | Y | | X509_issuer | ? |
> | Index_priv  | Y | | X509_subject| ? |
> | Alter_priv  | Y | | Max_questions   | 0 |
> | Show_db_priv| N | | Max_updates | 0 |
> | Super_priv  | N | | Max_connections | 0 |
> | Create_tmp_table_priv | Y |   | Max_user_connections | 0 |
> +-+---+ +-+---+
> NOTE:A password is required for user `opensips' :-(
> 
> The following rules are used:
>  db:
> '127.0.0.1','opensips','opensips','Y','Y','Y','Y','Y','Y','N','Y','Y','Y','Y','Y','Y','Y','Y','Y','Y'
>  host  : 'Not processed: host-field is not empty in db-table.'
>  user  :
> '%','opensips','*4DDB979A2666D0CF0A83FCCED820A64E8EBB6AFD','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','','','','','0','0','0','0'
> 
> 
> 
> On Tue, Jun 30, 2009 at 11:01 AM, Uwe Kastens  > wrote:
> 
> Hmm,
> 
> And you did change the host in the opensips url from localhost to
> 127.0.0.1?
> 
> For a test change the 127.0.0.1 in the mysql to % and try again. And
> mabye change the password with update user set
> password=password("yourpassword") where user="opensips";
> 
> 
> Your password entry looks like from a elder version.
> 
> BR
> 
> Uwe
> 
> 
> Brett Nemeroff schrieb:
> > Here is what I got:
> > password too long? really? it's not that long.. 
> >
> >
> > mysqlaccess Version 2.06, 20 Dec 2000
> > By RUG-AIV, by Yves Carlier (yves.carl...@rug.ac.be
> 
> > >)
> > Changes by Steve Harvey (s...@vex.net 
> >)
> > This software comes with ABSOLUTELY NO WARRANTY.
> > Password for MySQL superuser root:
> >
> > Access-rights
> > for USER 'opensips', from HOST 'localhost', to DB 'opensips'
> > +-+---+ +-+---+
> > | Select_priv | Y | | Lock_tables_priv | Y |
> > | Insert_priv | Y | | Execute_priv| Y |
> > | Update_priv | Y | | Repl_slave_priv | N |
> > | Delete_priv | Y | | Repl_client_priv | N |
> > | Create_priv | Y | | Create_view_priv | Y |
> > | Drop_priv   | Y | | Show_view_priv  | Y |
> > | Reload_priv | N | | Create_routine_priv | Y |
> > | Shutdown_priv   | N | | Alter_routine_priv | Y |
> > | Process_priv| N | | Create_user_priv | N |
> > | File_priv   | N | | Ssl_type| ? |
> > | Grant_priv  | N | | Ssl_cipher  | ? |
> > | References_priv | Y | | X509_issuer | ? |
> > | Index_priv  | Y | | X509_subject| ? |
> > | Alter_priv  | Y | | Max_questions   | 0 |
> > | Show_db_priv| N | | Max_updates | 0 |
> > | Super_priv  | N | | Max_connections | 0 |
> > | Create_tmp_table_priv | Y |   | Max_user_connections | 0 |
> > +-+---+ +-+---+
> > NOTE:A password is required for user `opensips' :-(
> >
> > The following rules are used:
> >  db:
> >
> 
> '127.0.0.1','opensips','opensips','Y','Y','Y','Y','Y','Y','N','Y','Y','Y','Y','Y','Y','Y','Y','Y','Y'
> >  host  : 'Not processed: host-field is not empty in db-table.'
> >  user  :
> >
> 
> '127.0.0.1','opensips','641b9f69397f5d64','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','','','','','0','0','0','0'
> >
> >
> > On Tue, Jun 30, 2009 at 10:41 AM, Uwe Kastens  
> > 

Re: [OpenSIPS-Users] Mediaproxy ver 2.3.4 - Conntrack meets Music on Hold

[Stuart Marsden]

3) detect this special case and change existing
conntrack to point to
  
the MoH sever, putting it back later
  

Ruud Klaver wrote:

How would you detect changes at SIP level in this case? The reason I'm
asking is because the proper way your phone should deal with this is by
sending a re-INVITE with the RTP endpoint of the MoH server in the SDP.
This should trigger mediaproxy into resetting the conntrack rule. Does
it currently do this, because this should already be working.


Stuart wrote:

The problem is the Sipura/Linksys/Cisco  Phones do it the other way
round - the phone going on hold  (A) sends a normal sendonly to the
caller(B) . Then (A) sends a brand new invite to the MoH server(C) ,
passing what it thinks is the  SDP of the phone (B), but is actually
the mediarelay

The 200OK SDP from the MoH sever is not used

So MoH starts to send Audio to what it thinks is (A) but actually sends
it to (B) 

I can detect at the SIP level the invite to MoH (based on the TO), the
SDP contains the IP:Port of a valid conntrack session , so inside
mediaproxy would need a new method called from opensips - "update the
conntrack session (ignoring callid,)  using SDP IP:Port as a key"
Ignoring the 200OK from the MoH server for now

Then when the user is taken off hold the phone (A) will send a new
invite to undo the sendonly to (B) and drop the call to (C)

Disruption to existing audio is fine as this will only happen on
transition to / from MoH

WDYT?

Stuart


Ruud Klaver wrote:
Hi Stuart,
  
  
On 30 Jun 2009, at 14:41, Stuart Marsden wrote:
  
  
  Hi,


I am testing a mediaproxy sever using 2.3.4


The tests I have done so far have worked, however there is one scenario

where once the call has been setup a phone can generate Music on Hold


It does this by making a 2nd call (unrelated to the first at the SIP

level) and passes what it thinks is  the IP:port of the 1st call far
end

(actually the mediaproxy) to the MoH server.


the MoH sever then sends audio to the mediaproxy.


However since the source of the audio is not the phone, conntrack is
not

forwarding on the audio to the other end.


Options as far as I can see are


1) remove the restriction on the source IP on who is allowed to forward

rtp - from what I have ready this probably wont work

  
  
The conntrack rule is bidirectional. If you allow data from multiple
sources to one destination then how would the relay know where to send
traffic in the reverse direction to?
  
  
  2) detect this special case and construct a
temporary additional  "1

way"  conntrack just for the MoH

  
  
No such thing, as they are bidirectional.
  
  
  3) detect this special case and change
existing conntrack to point to

the MoH sever, putting it back later

  
  
We actually considered implementing this, but not for this particular
scenario. Some modems seem to switch IP address every X hours. If this
change is during a relayed call, the RTP from one end will suddenly
come from a new IP address, like in your scenario. What could be done
is that the relay keeps listening on the allocated ports in userspace
for traffic from a different host and switch the conntrack rule to the
new IP if there is no traffic coming in anymore from the old IP.
  
  
The downside of this is that it could actually allow someone to hijack
your call if the relay and port are known (and there is a temporary
glitch in traffic in the conntrack rule).
  
  
  changes would be detected at the SIP level,
or more likely conntrack

call backs


any suggestions / help appreciated.


Stuart

  
  
How would you detect changes at SIP level in this case? The reason I'm
asking is because the proper way your phone should deal with this is by
sending a re-INVITE with the RTP endpoint of the MoH server in the SDP.
This should trigger mediaproxy into resetting the conntrack rule. Does
it currently do this, because this should already be working.
  
  
Ruud Klaver
  
AG Projects
  
  
  
  


-- 
_

Dr Stuart Marsden

MyPhones.com -  Extreme Digital Voice
Meade House
Chesham Rd
Wigginton
Tring
Herts
HP23 6JE

Myphones Voice: +44 (0)1494 414100

Legacy Voice: +44 (0)1494 758204
Fax: +44 (0)1494 757000

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mail to: stu...@myphones.com

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Re: [OpenSIPS-Users] ERROR:core:db_allocate_rows: no memory left - Please help

[mani sivaraman]

Great. I don't see that error any more after applying the patch.
Thanks

On Mon, Jun 29, 2009 at 12:01 PM, Bogdan-Andrei Iancu <
bog...@voice-system.ro> wrote:

> Hi Mani,
>
> if this will make the testing easier, here is the patch you can apply
> against 1.5.
>
> Regards,
> Bogdan
>
> mani sivaraman wrote:
>
>> Thanks Bogdan, I will definitely try the devel (1.6). I cannot install 1.6
>> on the current server I have. I have to setup another box and then try it
>> out. I will let you know. Mean while I'm trying hard to make xmpp2simple
>> presence status work. Any help or pointer would be appreciated.
>>
>> Thanks
>> Mani
>>
>> On Mon, Jun 29, 2009 at 11:04 AM, Bogdan-Andrei Iancu <
>> bog...@voice-system.ro > wrote:
>>
>>Hi Mani,
>>
>>I fixed this on 1.6 (devel) version by adding fetch support in the
>>PUA module, so now it can add as many row as possible without any
>>mem issues. Could you give it a try? if you confirm the fix, I
>>will do a backport to 1.5 also.
>>
>>Best regards,
>>Bogdan
>>
>>mani sivaraman wrote:
>>
>>I'm using opensips 1.5.1 for the past 1 week. Suddenly I get
>>this error below when starting. What is going on.
>>
>>
>>
>>Jun 26 15:52:14 [2535] ERROR:core:db_allocate_rows: no memory left
>>Jun 26 15:52:14 [2535] ERROR:db_mysql:db_mysql_convert_rows:
>>no private memory left
>>Jun 26 15:52:14 [2535] ERROR:db_mysql:db_mysql_convert_result:
>>error while converting rows
>>Jun 26 15:52:14 [2535] DBG:core:db_free_columns: freeing
>>result columns at 0x81a9b78
>>Jun 26 15:52:14 [2535] ERROR:db_mysql:db_mysql_store_result:
>>error while converting result
>>Jun 26 15:52:14 [2535] ERROR:core:db_do_query: error while
>>storing resultJun 26 15:52:14 [2535] ERROR:pua:db_restore:
>>while querrying table
>>Jun 26 15:52:14 [2535] ERROR:pua:mod_init: while restoring
>>hash_table
>>Jun 26 15:52:14 [2535] ERROR:core:init_mod: failed to
>>initialize module pua
>>Jun 26 15:52:14 [2535] ERROR:core:main: error while
>>initializing modules
>>Jun 26 15:52:14 [2535] DBG:xmpp:destroy: cleaning up...
>>Jun 26 15:52:14 [2535] DBG:pua:destroy: destroying module ...
>>Jun 26 15:52:14 [2535] DBG:db_mysql:db_mysql_submit_query:
>>discon reset for 135960296
>>Jun 26 15:52:14 [2535] DBG:core:pool_remove: removing
>>connection from the pool
>>Jun 26 15:52:14 [2535] DBG:presence_xml:destroy: start
>>Jun 26 15:52:14 [2535] NOTICE:presence:destroy: destroy module ...
>>Jun 26 15:52:14 [2535] DBG:xlog:destroy: destroy module...
>>Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: tm_shutdown : start
>>Jun 26 15:52:14 [2535] DBG:tm:unlink_timer_lists: emptying
>>DELETE list
>>Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: emptying hash table
>>Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: releasing timers
>>Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: removing semaphores
>>Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: destroying callback
>>lists
>>Jun 26 15:52:14 [2535] DBG:tm:tm_shutdown: tm_shutdown : done
>>Jun 26 15:52:14 [2535] DBG:core:destroy_tls: entered
>>Jun 26 15:52:14 [2535] DBG:core:shm_mem_destroy:
>>Jun 26 15:52:14 [2535] DBG:core:shm_mem_destroy: destroying
>>the shared memory lock
>>
>>
>>  
>>
>>
>>
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>>
>>
>>
>>
>
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Re: [OpenSIPS-Users] OpenSIPS-mediaproxy, Asterisk (packet2packet bridge) - problem with RTP

[Ruud Klaver]

Hi,

On 29 Jun 2009, at 08:14, Dimitrios Giannakopoulos wrote:

> Hi,
>
> I have implemented the following scenario:
>
> [incoming pstn]--->[opensips]-->[asterisk] --->[sip phone]
> |
> [outgoing pstn]<---[opensips]<--|
>
> Opensips acts as SBC with mediaproxy functionality. Moreover, I use
> the LCR module to route calls.
> The Asterisk is located at the public domain and we have activated the
> packet2packet bridge. A soft phone is registered to asterisk and we
> have created a ring group that sends an incoming call to soft phone
> and external line (outbound pstn) that rings simultaneous both
> devices. Opesips version 1.4.5 or 1.5 Asterisk version 1.6
>
> Single calls without ring gourp:
>
> Incoming calls from PSTN to asterisk through Opensips with mediaproxy
> enabled. It works properly.
> Outgoing calls from Asterisk to PSTN through Opensips with mediaproxy
> enabled. It works properly.
>
>
> Calls with ring group enabled:
> Incoming call from PSTN to asterisk through opensips with mediaproxy
> enabled. The incoming call activate the asterisk's  ring group and
> sends the call to sip phone and external line – outgoing pstn call.
> Both devices ring simultaneous. When hang-up:
>
> A) soft phone, the signaling and media work properly.
> B) External line, the signaling works properly but the media is not
> open. The system (opensips/mediaproxy) generates two media
> sessions(incoming and outgoing) but the ip of asterisk at both
> sessions has value Unknown. The mediaproxy/opensips tries to connect
> the two legs through asterisk. But this does not work because the
> asterisk acts as packet2packet bridge.
>
>
> Please, can you provide any help/sugestion about this problem?

Firstly I would strongly suggest that you only send your question  
once, sending it more than once will not get it answered sooner.

Secondly, I do not know what a Asterisk packet2packet bridge or a ring  
group is and have no intention of finding out.

Thirdly, I would suggest that you include logs from the relay from the  
last two scenarios you described.

Ruud Klaver
AG Projects
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Re: [OpenSIPS-Users] Parallel forking (using registrar branching) & mediaproxy

[Ruud Klaver]

Hi Erwan,

On 15 Jun 2009, at 18:09, erwan.hu...@orange-ftgroup.com wrote:

> Following your advice, i upgraded my platform to the following  
> architecture (including private IPs & NAT):
>
>  x
>   TDMx   x  10.0.100.2
> Phone 8003---x MGW 1 x--|
>  x   x  |
>  x  |
>  
> |xx
>  x  |   
> x xx
>   TDMx   x  10.0.100.3  |  10.0.100.1  x
> x  90.84.16.21   90.84.16.17  x  Opensips  x
> Phone 8003---x MGW 2 x--|--x  NAT   
> x-x Mediaproxy x
>  x   x  |  x
> x x inc. relay x
>  x  |   
> x xx
>  
> |xx
>  x  |
>  x   x  10.0.100.5  |
>  x  PC   x--|
>  x   x
>  x
>
> Note: 90.84.16.17 being the listening IP for both opensips &  
> mediaproxy.
>
> Using those NAT translation rules:
> -> 10.0.100.2 turned into 90.84.16.2
> -> 10.0.100.3 turned into 90.84.16.3
> -> 10.0.100.5 turned into 90.84.16.5

That's a much better setup than your previous one. I really can't tell  
you what a relay with two IPs when you're only listening on one does.

BTW, strictly speaking NATing to different IPs is not necessary, it  
should work when NATting all to the same IP, but it's a good  
approximation of a realistic situation.

> DISPATCHER / MGW2 user picks up (working)
> 
> debug: Connection from relay at 90.84.16.17
> debug: Issuing "sessions" command to relay at 90.84.16.17
> debug: Issuing "update" command to relay at 90.84.16.17
> debug: Issuing "update" command to relay at 90.84.16.17
> debug: Issuing "remove" command to relay at 90.84.16.17
> debug: Got statistics: {'from_tag':  
> 'E7ACBFABF6795C968B3F8EE4C65008E7', 'dialog_id': None, 'start_time':  
> 1245079081.8, 'timed_out': False, 'call_id': 
> 'e221500403491821a52e0c759cdf5...@cust1.sipetrali.fr 
> ', 'to_tag': '254B127C-C20', 'streams': [{'status': 'rejected',  
> 'caller_codec': 'Unknown', 'post_dial_delay': None, 'callee_codec':  
> 'Unknown', 'start_time': 0, 'caller_bytes': 0, 'callee_bytes': 0,  
> 'caller_packets': 0, 'end_time': 0, 'callee_remote': 'Unknown',  
> 'caller_remote': 'Unknown', 'media_type': 'video', 'callee_local':  
> '90.84.16.17:50006', 'timeout_wait': 0, 'caller_local':  
> '90.84.16.17:50004', 'callee_packets': 0}, {'status': 'closed',  
> 'caller_codec': 'Unknown(13)', 'post_dial_delay':  
> 0.5405640602111, 'callee_codec': 'G711u', 'start_time': 0,  
> 'caller_bytes': 59336, 'callee_bytes': 58800, 'caller_packets': 297,  
> 'end_time': 6, 'callee_remote': '90.84.16.3:19194', 'caller_remote':  
> '90.84.16.5:5166', 'media_type': 'audio', 'callee_local':  
> '90.84.16.17:50002', 'timeout_wait': 0, 'caller_local':  
> '90.84.16.17:5', 'callee_packets': 294}], 'duration': 6,  
> 'to_uri': '8...@90.84.16.17', 'from_uri': 'c...@cust1.sipetrali.fr',  
> 'callee_ua': 'Cisco-SIPGateway/IOS-12.x', 'caller_ua': 'Kapanga  
> Softphone Desktop 1.00/2172f 
> +1213709005_001CBFA9A746_001C233756B9_005056C1_005056C8'}
>
> DISPATCHER / MGW1 user picks up (NOT working)
> 
> debug: Issuing "update" command to relay at 90.84.16.17
> debug: Issuing "update" command to relay at 90.84.16.17
> debug: Issuing "remove" command to relay at 90.84.16.17
> debug: Got statistics: {'from_tag':  
> 'CA3D6F5E2FF9CDAC4585370AF309F62A', 'dialog_id': None, 'start_time':  
> 1245079099.141, 'timed_out': False, 'call_id': 
> '1e5ec56c5a7d8879c68fb1fa02d1a...@cust1.sipetrali.fr 
> ', 'to_tag': '1EEA090-37D', 'streams': [{'status': 'rejected',  
> 'caller_codec': 'Unknown', 'post_dial_delay': None, 'callee_codec':  
> 'Unknown', 'start_time': 0, 'caller_bytes': 0, 'callee_bytes': 0,  
> 'caller_packets': 0, 'end_time': 0, 'callee_remote': 'Unknown',  
> 'caller_remote': 'Unknown', 'media_type': 'video', 'callee_local':  
> '90.84.16.17:50014', 'timeout_wait': 0, 'caller_local':  
> '90.84.16.17:50012', 'callee_packets': 0}, {'status': 'closed',  
> 'caller_codec': 'Unknown(13)', 'post_dial_delay':  
> 0.5387377738949, 'callee_codec': 'G711u', 'start_time': 0,  
> 'caller_bytes': 55800, 'callee_bytes': 0, 'caller_packets': 279,  
> 'end_time': 5, 'callee_remote': '90.84.16.3:17994', 'caller_remote':  
>

Re: [OpenSIPS-Users] mysql database connection error

[Brett Nemeroff]

Still can't connect :( What could I be doing wrong? I wonder if it's the
mysql client libs somehow?!
Access-rights
for USER 'opensips', from HOST 'localhost', to DB 'opensips'
+-+---+ +-+---+
| Select_priv | Y | | Lock_tables_priv | Y |
| Insert_priv | Y | | Execute_priv| Y |
| Update_priv | Y | | Repl_slave_priv | N |
| Delete_priv | Y | | Repl_client_priv | N |
| Create_priv | Y | | Create_view_priv | Y |
| Drop_priv   | Y | | Show_view_priv  | Y |
| Reload_priv | N | | Create_routine_priv | Y |
| Shutdown_priv   | N | | Alter_routine_priv | Y |
| Process_priv| N | | Create_user_priv | N |
| File_priv   | N | | Ssl_type| ? |
| Grant_priv  | N | | Ssl_cipher  | ? |
| References_priv | Y | | X509_issuer | ? |
| Index_priv  | Y | | X509_subject| ? |
| Alter_priv  | Y | | Max_questions   | 0 |
| Show_db_priv| N | | Max_updates | 0 |
| Super_priv  | N | | Max_connections | 0 |
| Create_tmp_table_priv | Y |   | Max_user_connections | 0 |
+-+---+ +-+---+
NOTE:A password is required for user `opensips' :-(

The following rules are used:
 db:
'127.0.0.1','opensips','opensips','Y','Y','Y','Y','Y','Y','N','Y','Y','Y','Y','Y','Y','Y','Y','Y','Y'
 host  : 'Not processed: host-field is not empty in db-table.'
 user  :
'%','opensips','*4DDB979A2666D0CF0A83FCCED820A64E8EBB6AFD','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','','','','','0','0','0','0'



On Tue, Jun 30, 2009 at 11:01 AM, Uwe Kastens  wrote:

> Hmm,
>
> And you did change the host in the opensips url from localhost to
> 127.0.0.1?
>
> For a test change the 127.0.0.1 in the mysql to % and try again. And
> mabye change the password with update user set
> password=password("yourpassword") where user="opensips";
>
>
> Your password entry looks like from a elder version.
>
> BR
>
> Uwe
>
>
> Brett Nemeroff schrieb:
> > Here is what I got:
> > password too long? really? it's not that long.. 
> >
> >
> > mysqlaccess Version 2.06, 20 Dec 2000
> > By RUG-AIV, by Yves Carlier (yves.carl...@rug.ac.be
> > )
> > Changes by Steve Harvey (s...@vex.net )
> > This software comes with ABSOLUTELY NO WARRANTY.
> > Password for MySQL superuser root:
> >
> > Access-rights
> > for USER 'opensips', from HOST 'localhost', to DB 'opensips'
> > +-+---+ +-+---+
> > | Select_priv | Y | | Lock_tables_priv | Y |
> > | Insert_priv | Y | | Execute_priv| Y |
> > | Update_priv | Y | | Repl_slave_priv | N |
> > | Delete_priv | Y | | Repl_client_priv | N |
> > | Create_priv | Y | | Create_view_priv | Y |
> > | Drop_priv   | Y | | Show_view_priv  | Y |
> > | Reload_priv | N | | Create_routine_priv | Y |
> > | Shutdown_priv   | N | | Alter_routine_priv | Y |
> > | Process_priv| N | | Create_user_priv | N |
> > | File_priv   | N | | Ssl_type| ? |
> > | Grant_priv  | N | | Ssl_cipher  | ? |
> > | References_priv | Y | | X509_issuer | ? |
> > | Index_priv  | Y | | X509_subject| ? |
> > | Alter_priv  | Y | | Max_questions   | 0 |
> > | Show_db_priv| N | | Max_updates | 0 |
> > | Super_priv  | N | | Max_connections | 0 |
> > | Create_tmp_table_priv | Y |   | Max_user_connections | 0 |
> > +-+---+ +-+---+
> > NOTE:A password is required for user `opensips' :-(
> >
> > The following rules are used:
> >  db:
> >
> '127.0.0.1','opensips','opensips','Y','Y','Y','Y','Y','Y','N','Y','Y','Y','Y','Y','Y','Y','Y','Y','Y'
> >  host  : 'Not processed: host-field is not empty in db-table.'
> >  user  :
> >
> '127.0.0.1','opensips','641b9f69397f5d64','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','','','','','0','0','0','0'
> >
> >
> > On Tue, Jun 30, 2009 at 10:41 AM, Uwe Kastens  > > wrote:
> >
> > Hi,
> >
> > Hmmm.
> > - password to long
> >
> > Have you mysqlaccess on the system?
> > This is how the output looks on my testsystem
> >
> >
> >
> >  mysqlaccess  localhost opensips opensips -U root -P
> > mysqlaccess Version 2.06, 20 Dec 2000
> > By RUG-AIV, by Yves Carlier (yves.carl...@rug.ac.be
> > )
> > Changes by Steve Harvey (s...@vex.net )
> > This software comes with ABSOLUTELY NO WARRANTY.
> > Password for MySQL superuser root:
> >
> > Access-rights
> > for USER 'opensips', from HOST 'localhost', to DB 'ope

[OpenSIPS-Users] [Webinar] Next webinar is "Routing is SIP"

[Bogdan-Andrei Iancu]

Hi,

The next free webinar is scheduled for Wednesday 8th of July 2009. The 
topic is "Routing in SIP"

This webinar will help you to understand SIP routing concepts. SIP 
routing is very often a misunderstood concept. It is essential for 
anyone trying to script OpenSIPS to fully understand these concepts.

Registration form : https://www2.gotomeeting.com/register/429289130

Also see http://www.opensips.org/Training/Webinars for future updates.

Best regards,
Bogdan



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Re: [OpenSIPS-Users] Mediaproxy ver 2.3.4 - Conntrack meets Music on Hold

[Ruud Klaver]

Hi Stuart,

On 30 Jun 2009, at 14:41, Stuart Marsden wrote:

> Hi,
>
> I am testing a mediaproxy sever using 2.3.4
>
> The tests I have done so far have worked, however there is one  
> scenario
> where once the call has been setup a phone can generate Music on Hold
>
> It does this by making a 2nd call (unrelated to the first at the SIP
> level) and passes what it thinks is  the IP:port of the 1st call far  
> end
> (actually the mediaproxy) to the MoH server.
>
> the MoH sever then sends audio to the mediaproxy.
>
> However since the source of the audio is not the phone, conntrack is  
> not
> forwarding on the audio to the other end.
>
> Options as far as I can see are
>
> 1) remove the restriction on the source IP on who is allowed to  
> forward
> rtp - from what I have ready this probably wont work

The conntrack rule is bidirectional. If you allow data from multiple  
sources to one destination then how would the relay know where to send  
traffic in the reverse direction to?

> 2) detect this special case and construct a temporary additional  "1
> way"  conntrack just for the MoH

No such thing, as they are bidirectional.

> 3) detect this special case and change existing conntrack to point to
> the MoH sever, putting it back later

We actually considered implementing this, but not for this particular  
scenario. Some modems seem to switch IP address every X hours. If this  
change is during a relayed call, the RTP from one end will suddenly  
come from a new IP address, like in your scenario. What could be done  
is that the relay keeps listening on the allocated ports in userspace  
for traffic from a different host and switch the conntrack rule to the  
new IP if there is no traffic coming in anymore from the old IP.

The downside of this is that it could actually allow someone to hijack  
your call if the relay and port are known (and there is a temporary  
glitch in traffic in the conntrack rule).

> changes would be detected at the SIP level, or more likely conntrack
> call backs
>
> any suggestions / help appreciated.
>
> Stuart

How would you detect changes at SIP level in this case? The reason I'm  
asking is because the proper way your phone should deal with this is  
by sending a re-INVITE with the RTP endpoint of the MoH server in the  
SDP. This should trigger mediaproxy into resetting the conntrack rule.  
Does it currently do this, because this should already be working.

Ruud Klaver
AG Projects


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Re: [OpenSIPS-Users] mysql database connection error

[Uwe Kastens]

Hmm,

And you did change the host in the opensips url from localhost to
127.0.0.1?

For a test change the 127.0.0.1 in the mysql to % and try again. And
mabye change the password with update user set
password=password("yourpassword") where user="opensips";


Your password entry looks like from a elder version.

BR

Uwe


Brett Nemeroff schrieb:
> Here is what I got:
> password too long? really? it's not that long.. 
> 
> 
> mysqlaccess Version 2.06, 20 Dec 2000
> By RUG-AIV, by Yves Carlier (yves.carl...@rug.ac.be
> )
> Changes by Steve Harvey (s...@vex.net )
> This software comes with ABSOLUTELY NO WARRANTY.
> Password for MySQL superuser root: 
> 
> Access-rights
> for USER 'opensips', from HOST 'localhost', to DB 'opensips'
> +-+---+ +-+---+
> | Select_priv | Y | | Lock_tables_priv | Y |
> | Insert_priv | Y | | Execute_priv| Y |
> | Update_priv | Y | | Repl_slave_priv | N |
> | Delete_priv | Y | | Repl_client_priv | N |
> | Create_priv | Y | | Create_view_priv | Y |
> | Drop_priv   | Y | | Show_view_priv  | Y |
> | Reload_priv | N | | Create_routine_priv | Y |
> | Shutdown_priv   | N | | Alter_routine_priv | Y |
> | Process_priv| N | | Create_user_priv | N |
> | File_priv   | N | | Ssl_type| ? |
> | Grant_priv  | N | | Ssl_cipher  | ? |
> | References_priv | Y | | X509_issuer | ? |
> | Index_priv  | Y | | X509_subject| ? |
> | Alter_priv  | Y | | Max_questions   | 0 |
> | Show_db_priv| N | | Max_updates | 0 |
> | Super_priv  | N | | Max_connections | 0 |
> | Create_tmp_table_priv | Y |   | Max_user_connections | 0 |
> +-+---+ +-+---+
> NOTE:A password is required for user `opensips' :-(
> 
> The following rules are used:
>  db:
> '127.0.0.1','opensips','opensips','Y','Y','Y','Y','Y','Y','N','Y','Y','Y','Y','Y','Y','Y','Y','Y','Y'
>  host  : 'Not processed: host-field is not empty in db-table.'
>  user  :
> '127.0.0.1','opensips','641b9f69397f5d64','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','','','','','0','0','0','0'
> 
> 
> On Tue, Jun 30, 2009 at 10:41 AM, Uwe Kastens  > wrote:
> 
> Hi,
> 
> Hmmm.
> - password to long
> 
> Have you mysqlaccess on the system?
> This is how the output looks on my testsystem
> 
> 
> 
>  mysqlaccess  localhost opensips opensips -U root -P
> mysqlaccess Version 2.06, 20 Dec 2000
> By RUG-AIV, by Yves Carlier (yves.carl...@rug.ac.be
> )
> Changes by Steve Harvey (s...@vex.net )
> This software comes with ABSOLUTELY NO WARRANTY.
> Password for MySQL superuser root:
> 
> Access-rights
> for USER 'opensips', from HOST 'localhost', to DB 'opensips'
>+-+---+ +-+---+
>| Select_priv | Y | | Lock_tables_priv | Y |
>| Insert_priv | Y | | Execute_priv| Y |
>| Update_priv | Y | | Repl_slave_priv | N |
>| Delete_priv | Y | | Repl_client_priv | N |
>| Create_priv | Y | | Create_view_priv | Y |
>| Drop_priv   | Y | | Show_view_priv  | Y |
>| Reload_priv | N | | Create_routine_priv | Y |
>| Shutdown_priv   | N | | Alter_routine_priv | Y |
>| Process_priv| N | | Create_user_priv | N |
>| File_priv   | N | | Ssl_type| ? |
>| Grant_priv  | N | | Ssl_cipher  | ? |
>| References_priv | Y | | X509_issuer | ? |
>| Index_priv  | Y | | X509_subject| ? |
>| Alter_priv  | Y | | Max_questions   | 0 |
>| Show_db_priv| N | | Max_updates | 0 |
>| Super_priv  | N | | Max_connections | 0 |
>| Create_tmp_table_priv | Y |   | Max_user_connections | 0 |
>+-+---+ +-+---+
> NOTE:A password is required for user `opensips' :-(
> 
> The following rules are used:
>  db:
> 
> '%','opensips','opensips','Y','Y','Y','Y','Y','Y','N','Y','Y','Y','Y','Y','Y','Y','Y','Y','Y'
>  host  : 'Not processed: host-field is not empty in db-table.'
>  user  :
> 
> '%','opensips','*30C28A928E2BE5EFD59FF20CB8705B31ACCF3730','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','','','','','0','0','0','0'
> 
> BUGs can be reported by email to b...@mysql.com 
> 
> 
> BR
> 
> Uwe
> 
> 
> Brett Nemeroff schrieb:
> > Yeah, it's in there. I'm really puzzled. This should be the easy
> part. ;)
>

Re: [OpenSIPS-Users] mysql database connection error

[Brett Nemeroff]

Here is what I got:password too long? really? it's not that long.. 


mysqlaccess Version 2.06, 20 Dec 2000
By RUG-AIV, by Yves Carlier (yves.carl...@rug.ac.be)
Changes by Steve Harvey (s...@vex.net)
This software comes with ABSOLUTELY NO WARRANTY.
Password for MySQL superuser root:

Access-rights
for USER 'opensips', from HOST 'localhost', to DB 'opensips'
+-+---+ +-+---+
| Select_priv | Y | | Lock_tables_priv | Y |
| Insert_priv | Y | | Execute_priv| Y |
| Update_priv | Y | | Repl_slave_priv | N |
| Delete_priv | Y | | Repl_client_priv | N |
| Create_priv | Y | | Create_view_priv | Y |
| Drop_priv   | Y | | Show_view_priv  | Y |
| Reload_priv | N | | Create_routine_priv | Y |
| Shutdown_priv   | N | | Alter_routine_priv | Y |
| Process_priv| N | | Create_user_priv | N |
| File_priv   | N | | Ssl_type| ? |
| Grant_priv  | N | | Ssl_cipher  | ? |
| References_priv | Y | | X509_issuer | ? |
| Index_priv  | Y | | X509_subject| ? |
| Alter_priv  | Y | | Max_questions   | 0 |
| Show_db_priv| N | | Max_updates | 0 |
| Super_priv  | N | | Max_connections | 0 |
| Create_tmp_table_priv | Y |   | Max_user_connections | 0 |
+-+---+ +-+---+
NOTE:A password is required for user `opensips' :-(

The following rules are used:
 db:
'127.0.0.1','opensips','opensips','Y','Y','Y','Y','Y','Y','N','Y','Y','Y','Y','Y','Y','Y','Y','Y','Y'
 host  : 'Not processed: host-field is not empty in db-table.'
 user  :
'127.0.0.1','opensips','641b9f69397f5d64','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','','','','','0','0','0','0'


On Tue, Jun 30, 2009 at 10:41 AM, Uwe Kastens  wrote:

> Hi,
>
> Hmmm.
> - password to long
>
> Have you mysqlaccess on the system?
> This is how the output looks on my testsystem
>
>
>
>  mysqlaccess  localhost opensips opensips -U root -P
> mysqlaccess Version 2.06, 20 Dec 2000
> By RUG-AIV, by Yves Carlier (yves.carl...@rug.ac.be)
> Changes by Steve Harvey (s...@vex.net)
> This software comes with ABSOLUTELY NO WARRANTY.
> Password for MySQL superuser root:
>
> Access-rights
> for USER 'opensips', from HOST 'localhost', to DB 'opensips'
>+-+---+ +-+---+
>| Select_priv | Y | | Lock_tables_priv | Y |
>| Insert_priv | Y | | Execute_priv| Y |
>| Update_priv | Y | | Repl_slave_priv | N |
>| Delete_priv | Y | | Repl_client_priv | N |
>| Create_priv | Y | | Create_view_priv | Y |
>| Drop_priv   | Y | | Show_view_priv  | Y |
>| Reload_priv | N | | Create_routine_priv | Y |
>| Shutdown_priv   | N | | Alter_routine_priv | Y |
>| Process_priv| N | | Create_user_priv | N |
>| File_priv   | N | | Ssl_type| ? |
>| Grant_priv  | N | | Ssl_cipher  | ? |
>| References_priv | Y | | X509_issuer | ? |
>| Index_priv  | Y | | X509_subject| ? |
>| Alter_priv  | Y | | Max_questions   | 0 |
>| Show_db_priv| N | | Max_updates | 0 |
>| Super_priv  | N | | Max_connections | 0 |
>| Create_tmp_table_priv | Y |   | Max_user_connections | 0 |
>+-+---+ +-+---+
> NOTE:A password is required for user `opensips' :-(
>
> The following rules are used:
>  db:
>
> '%','opensips','opensips','Y','Y','Y','Y','Y','Y','N','Y','Y','Y','Y','Y','Y','Y','Y','Y','Y'
>  host  : 'Not processed: host-field is not empty in db-table.'
>  user  :
>
> '%','opensips','*30C28A928E2BE5EFD59FF20CB8705B31ACCF3730','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','','','','','0','0','0','0'
>
> BUGs can be reported by email to b...@mysql.com
>
>
> BR
>
> Uwe
>
>
> Brett Nemeroff schrieb:
> > Yeah, it's in there. I'm really puzzled. This should be the easy part. ;)
> >
> > Any other ideas?
> >
> >
> >
> > On Tue, Jun 30, 2009 at 10:25 AM, Uwe Kastens  > > wrote:
> >
> > Hi,
> >
> > looks good for me. Did you reload mysql after grant or flush priv?
> >
> > Should be, otherwise you wont be able to connect via mysql client.
> >
> > Your /etc/hosts have the entry for localhost?
> >
> > BR
> >
> > Uwe
> >
> > Brett Nemeroff schrieb:
> > > yeah, I tried localhost, 127.0.0.1, and the actual ip (I usually
> use
> > > localhost)
> > >
> > > here's my connect string:
> > > modparam("auth_db", "db_url",
> > > "mysql://opensips:23u83fwhw...@localhost/opensips")
> > >
> > > Here's the mysql grant:
> > > GRANT ALL ON opensips.* TO 'opensips'@'localhost' IDENTIFIED BY
> >  

Re: [OpenSIPS-Users] mysql database connection error

[Uwe Kastens]

Hi,

Hmmm.
- password to long

Have you mysqlaccess on the system?
This is how the output looks on my testsystem



 mysqlaccess  localhost opensips opensips -U root -P
mysqlaccess Version 2.06, 20 Dec 2000
By RUG-AIV, by Yves Carlier (yves.carl...@rug.ac.be)
Changes by Steve Harvey (s...@vex.net)
This software comes with ABSOLUTELY NO WARRANTY.
Password for MySQL superuser root:

Access-rights
for USER 'opensips', from HOST 'localhost', to DB 'opensips'
+-+---+ +-+---+
| Select_priv | Y | | Lock_tables_priv | Y |
| Insert_priv | Y | | Execute_priv| Y |
| Update_priv | Y | | Repl_slave_priv | N |
| Delete_priv | Y | | Repl_client_priv | N |
| Create_priv | Y | | Create_view_priv | Y |
| Drop_priv   | Y | | Show_view_priv  | Y |
| Reload_priv | N | | Create_routine_priv | Y |
| Shutdown_priv   | N | | Alter_routine_priv | Y |
| Process_priv| N | | Create_user_priv | N |
| File_priv   | N | | Ssl_type| ? |
| Grant_priv  | N | | Ssl_cipher  | ? |
| References_priv | Y | | X509_issuer | ? |
| Index_priv  | Y | | X509_subject| ? |
| Alter_priv  | Y | | Max_questions   | 0 |
| Show_db_priv| N | | Max_updates | 0 |
| Super_priv  | N | | Max_connections | 0 |
| Create_tmp_table_priv | Y |   | Max_user_connections | 0 |
+-+---+ +-+---+
NOTE:A password is required for user `opensips' :-(

The following rules are used:
 db:
'%','opensips','opensips','Y','Y','Y','Y','Y','Y','N','Y','Y','Y','Y','Y','Y','Y','Y','Y','Y'
 host  : 'Not processed: host-field is not empty in db-table.'
 user  :
'%','opensips','*30C28A928E2BE5EFD59FF20CB8705B31ACCF3730','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','N','','','','','0','0','0','0'

BUGs can be reported by email to b...@mysql.com


BR

Uwe


Brett Nemeroff schrieb:
> Yeah, it's in there. I'm really puzzled. This should be the easy part. ;)
> 
> Any other ideas?
> 
> 
> 
> On Tue, Jun 30, 2009 at 10:25 AM, Uwe Kastens  > wrote:
> 
> Hi,
> 
> looks good for me. Did you reload mysql after grant or flush priv?
> 
> Should be, otherwise you wont be able to connect via mysql client.
> 
> Your /etc/hosts have the entry for localhost?
> 
> BR
> 
> Uwe
> 
> Brett Nemeroff schrieb:
> > yeah, I tried localhost, 127.0.0.1, and the actual ip (I usually use
> > localhost)
> >
> > here's my connect string:
> > modparam("auth_db", "db_url",
> > "mysql://opensips:23u83fwhw...@localhost/opensips")
> >
> > Here's the mysql grant:
> > GRANT ALL ON opensips.* TO 'opensips'@'localhost' IDENTIFIED BY
> > '23u83fwhwkgh';
> >
> > On Tue, Jun 30, 2009 at 9:56 AM, Uwe Kastens  
> > >> wrote:
> >
> > Brett,
> >
> > Could you post your URL from your config please?
> >
> > 127.0.0.1 is not the same as localhost!
> > BR
> >
> > Uwe
> >
> > > Hey all,
> > > sorry for such a noob question here, but I just can't figure
> out what
> > > I'm doing wrong.. I'm getting the error:
> > > Jun 30 15:36:33 nguenj297 /usr/local/sbin/opensips[10159]:
> > > ERROR:db_mysql:db_mysql_new_connection: driver error(1045):
> Access
> > > denied for user 'opensips'@'localhost' (using password: YES)
> > >
> > > So of course, I checked the usernames and passwords.. I tried
> > logging in
> > > manually with:
> > > mysql -u opensips -h localhost -p opensips
> > > (with the same password of course)
> > >
> > > And it works fine.. so I'm not sure where the hangup is.. Where
> > should I
> > > debug this?
> > >
> > > Thanks,
> > > Brett
> > >
> > >
> > >
> > >
> > >
> > >
> >
> 
> > >
> > > ___
> > > Users mailing list
> > > Users@lists.opensips.org 
> >
> > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
> > --
> >
> > kiste lat: 54.322684, lon: 10.13586
> >
> >
> 
> 
> --
> 
> kiste lat: 54.322684, lon: 10.13586
> 
> 


-- 

kiste lat: 54.322684, lon: 10.13586

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Re: [OpenSIPS-Users] mysql database connection error

[Brett Nemeroff]

Yeah, it's in there. I'm really puzzled. This should be the easy part. ;)
Any other ideas?



On Tue, Jun 30, 2009 at 10:25 AM, Uwe Kastens  wrote:

> Hi,
>
> looks good for me. Did you reload mysql after grant or flush priv?
>
> Should be, otherwise you wont be able to connect via mysql client.
>
> Your /etc/hosts have the entry for localhost?
>
> BR
>
> Uwe
>
> Brett Nemeroff schrieb:
> > yeah, I tried localhost, 127.0.0.1, and the actual ip (I usually use
> > localhost)
> >
> > here's my connect string:
> > modparam("auth_db", "db_url",
> > "mysql://opensips:23u83fwhw...@localhost/opensips")
> >
> > Here's the mysql grant:
> > GRANT ALL ON opensips.* TO 'opensips'@'localhost' IDENTIFIED BY
> > '23u83fwhwkgh';
> >
> > On Tue, Jun 30, 2009 at 9:56 AM, Uwe Kastens  > > wrote:
> >
> > Brett,
> >
> > Could you post your URL from your config please?
> >
> > 127.0.0.1 is not the same as localhost!
> > BR
> >
> > Uwe
> >
> > > Hey all,
> > > sorry for such a noob question here, but I just can't figure out
> what
> > > I'm doing wrong.. I'm getting the error:
> > > Jun 30 15:36:33 nguenj297 /usr/local/sbin/opensips[10159]:
> > > ERROR:db_mysql:db_mysql_new_connection: driver error(1045): Access
> > > denied for user 'opensips'@'localhost' (using password: YES)
> > >
> > > So of course, I checked the usernames and passwords.. I tried
> > logging in
> > > manually with:
> > > mysql -u opensips -h localhost -p opensips
> > > (with the same password of course)
> > >
> > > And it works fine.. so I'm not sure where the hangup is.. Where
> > should I
> > > debug this?
> > >
> > > Thanks,
> > > Brett
> > >
> > >
> > >
> > >
> > >
> > >
> >
> 
> > >
> > > ___
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> > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> >
> > --
> >
> > kiste lat: 54.322684, lon: 10.13586
> >
> >
>
>
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>
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Re: [OpenSIPS-Users] opensips+dispatcher+asterisk problem

[ram]

On Tue, Jun 30, 2009 at 5:20 PM, Bogdan-Andrei Iancu  wrote:

> Hi Ram,
>
> I found your email on the Asterisk mailing list also ;)
>
> So, to answer here also: do you get any reply back from Asterisk ?
>

Hi Bogdan

thanks for the reply

I have made a quick Fix, iam not sure how far its good.

Just put coment in  secret , in the Asterisk Additional_a2billing_sip.conf.
rather doing twise  authentication.


But i have another problem here with the Dispatcher,
dispatcher sending calls round robin,

1 st call to 1st *
2nd call to 2nd *
3 call to 3rd *

if 2nd Asterisk fails to respond still Dispatcher module sending calls to
2nd asterisk

how can i fix this issue with Dispatcher, if any one of * box not reachable
it should detect and send call to 3rd *

if 2nd comes back in to network and live, it should send to 2nd *

how can i achive this ?

Ram
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Re: [OpenSIPS-Users] mysql database connection error

[Uwe Kastens]

Brett,

Could you post your URL from your config please?

127.0.0.1 is not the same as localhost!
BR

Uwe

> Hey all,
> sorry for such a noob question here, but I just can't figure out what
> I'm doing wrong.. I'm getting the error:
> Jun 30 15:36:33 nguenj297 /usr/local/sbin/opensips[10159]:
> ERROR:db_mysql:db_mysql_new_connection: driver error(1045): Access
> denied for user 'opensips'@'localhost' (using password: YES) 
> 
> So of course, I checked the usernames and passwords.. I tried logging in
> manually with:
> mysql -u opensips -h localhost -p opensips
> (with the same password of course) 
> 
> And it works fine.. so I'm not sure where the hangup is.. Where should I
> debug this?
> 
> Thanks,
> Brett
> 
> 
> 
> 
> 
> 
> 
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[OpenSIPS-Users] mysql database connection error

[Brett Nemeroff]

Hey all,sorry for such a noob question here, but I just can't figure out
what I'm doing wrong.. I'm getting the error:
Jun 30 15:36:33 nguenj297 /usr/local/sbin/opensips[10159]:
ERROR:db_mysql:db_mysql_new_connection: driver error(1045): Access denied
for user 'opensips'@'localhost' (using password: YES)

So of course, I checked the usernames and passwords.. I tried logging in
manually with:
mysql -u opensips -h localhost -p opensips
(with the same password of course)

And it works fine.. so I'm not sure where the hangup is.. Where should I
debug this?

Thanks,
Brett
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[OpenSIPS-Users] src ip check on Register => Re: How to insert the IP address of user in radius request.

[Uwe Kastens]

Hi,

this is the script part, that is doing the job. ATM just only logging

loadmodule "avpops.so"

#  avpops
modparam("avpops", "db_url","mysql://:x...@abcd.domain.de/testdb")

 if (method=="REGISTER") {
if (!radius_www_authorize("")) {
 www_challenge("", "0");
 exit;
 };
avp_db_query("select ip from src_ip where number='$au'",
"$avp(s:srcip)");
if ($avp(s:srcip)!=$si){
 xlog("$au should have SRC_IP $avp(s:srcip), but has $si");
}
 save("location") ;
 exit;
 }

BR

Uwe


Uwe Kastens schrieb:
> Hi,
> 
> I am facing a similar situation. We need to verify that a REGISTER comes
> from the same srcip we have configured in our database. I am thinking
> about doing this by making a select into an AVP and verfying the value
> of the AVP with the $si. If this is successfull the UA would be saved
> into the location and/or would be able to make a call.
> 
> This should be possible with radius_avp as well.
> 
> Looking at performance I would make the DIGEST Auth 1st and if this is
> succesfull check the IPs.
> 
> BR
> 
> uwe
> 
> 
> Tung Tran schrieb:
>> Hi Mr. Bogdan
>>
>> We need it for IP authorize besides DIGEST auth, that is not standard anyway 
>> but business requirements.
>> We use MSSQL to do DIGEST authorize and we need an extra security layer 
>> based on source IP, that is also a request by govements in my contry.
>>
>> So last but not lease, I would like someone can help me how to add this 
>> feature as soon ass possible
>>
>> Thank you very much for your help
>>
>> Tung
>> - Original Message - 
>> From: "Bogdan-Andrei Iancu" 
>> To: "Tung Tran" 
>> Cc: 
>> Sent: Friday, June 26, 2009 2:24 AM
>> Subject: Re: [OpenSIPS-Users] How to insert the IP address of user in radius 
>> request.
>>
>>
>>> Hi Tung,
>>>
>>> I see the difference - unfortunately there is no way (at the moment) to 
>>> add custom info to the RADIUS auth header, but it should be an extension 
>>> that can be done - out of curiosity? why do you need this in the AUTH 
>>> request, as this info is not used in the DIGEST auth.
>>>
>>> Regards,
>>> Bogdan
>>>
>>> Tung Tran wrote:
 Dear Mr. Bogdan,

 I know we can insert the source IP address in account request before 
 sending it to Radius, however can I insert it in AUTHORIZE request 
 instead?

 Thank you very much for your reply.
 Tung

 - Original Message - From: "Bogdan-Andrei Iancu" 
 
 To: "Tung Tran" 
 Cc: 
 Sent: Tuesday, June 23, 2009 6:04 PM
 Subject: Re: [OpenSIPS-Users] How to insert the IP address of user in 
 radius request.


> Hi Tung,
>
> First of all you should upgrade to 1.5 version (see 
> http://www.opensips.org/Resources/Downloads).
>
> For your problem, use extra accounting - you can account whatever extra 
> info you want. See:
>
> http://www.opensips.org/html/docs/modules/1.5.x/acc.html#ACC-extra-id
>
> To get the source IP, use the $si pseudo-variable (see 
> http://www.opensips.org/Resources/DocsCoreVar15#toc71).
>
> Regards,
> Bogdan
>
> Tung Tran wrote:
>> Hi all,
>>
>> I get a request to insert the public IP address of the sip softphone or 
>> IP Phone/ATA (end-point) in the Radius request sending to Radius 
>> server.
>> I am thinking about to mod the auth_radius module to insert that IP in 
>> SIP-URI-User field, likely this one:
>>
>> Original
>> Sip-Uri-User = "985512405"
>>
>> After mod:
>> Sip-Uri-User = 985512...@1.2.3.4
>>
>> Where 1.2.3.4 is the IP of SIP end-point, not the IP address of 
>> Opensips/Opensers servers.
>>
>> But I dont know where I should play with.
>> Any one had done it before or know where we can edit, pls help  me.
>>
>> BTW, I am using openser 1.2.2 version.
>> Thanks in advance
>> Tung
>>
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> 
> 


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Re: [OpenSIPS-Users] How to insert the IP address of user in radius request.

[Uwe Kastens]

Hi,

I am facing a similar situation. We need to verify that a REGISTER comes
from the same srcip we have configured in our database. I am thinking
about doing this by making a select into an AVP and verfying the value
of the AVP with the $si. If this is successfull the UA would be saved
into the location and/or would be able to make a call.

This should be possible with radius_avp as well.

Looking at performance I would make the DIGEST Auth 1st and if this is
succesfull check the IPs.

BR

uwe


Tung Tran schrieb:
> Hi Mr. Bogdan
> 
> We need it for IP authorize besides DIGEST auth, that is not standard anyway 
> but business requirements.
> We use MSSQL to do DIGEST authorize and we need an extra security layer 
> based on source IP, that is also a request by govements in my contry.
> 
> So last but not lease, I would like someone can help me how to add this 
> feature as soon ass possible
> 
> Thank you very much for your help
> 
> Tung
> - Original Message - 
> From: "Bogdan-Andrei Iancu" 
> To: "Tung Tran" 
> Cc: 
> Sent: Friday, June 26, 2009 2:24 AM
> Subject: Re: [OpenSIPS-Users] How to insert the IP address of user in radius 
> request.
> 
> 
>> Hi Tung,
>>
>> I see the difference - unfortunately there is no way (at the moment) to 
>> add custom info to the RADIUS auth header, but it should be an extension 
>> that can be done - out of curiosity? why do you need this in the AUTH 
>> request, as this info is not used in the DIGEST auth.
>>
>> Regards,
>> Bogdan
>>
>> Tung Tran wrote:
>>> Dear Mr. Bogdan,
>>>
>>> I know we can insert the source IP address in account request before 
>>> sending it to Radius, however can I insert it in AUTHORIZE request 
>>> instead?
>>>
>>> Thank you very much for your reply.
>>> Tung
>>>
>>> - Original Message - From: "Bogdan-Andrei Iancu" 
>>> 
>>> To: "Tung Tran" 
>>> Cc: 
>>> Sent: Tuesday, June 23, 2009 6:04 PM
>>> Subject: Re: [OpenSIPS-Users] How to insert the IP address of user in 
>>> radius request.
>>>
>>>
 Hi Tung,

 First of all you should upgrade to 1.5 version (see 
 http://www.opensips.org/Resources/Downloads).

 For your problem, use extra accounting - you can account whatever extra 
 info you want. See:

 http://www.opensips.org/html/docs/modules/1.5.x/acc.html#ACC-extra-id

 To get the source IP, use the $si pseudo-variable (see 
 http://www.opensips.org/Resources/DocsCoreVar15#toc71).

 Regards,
 Bogdan

 Tung Tran wrote:
> Hi all,
>
> I get a request to insert the public IP address of the sip softphone or 
> IP Phone/ATA (end-point) in the Radius request sending to Radius 
> server.
> I am thinking about to mod the auth_radius module to insert that IP in 
> SIP-URI-User field, likely this one:
>
> Original
> Sip-Uri-User = "985512405"
>
> After mod:
> Sip-Uri-User = 985512...@1.2.3.4
>
> Where 1.2.3.4 is the IP of SIP end-point, not the IP address of 
> Opensips/Opensers servers.
>
> But I dont know where I should play with.
> Any one had done it before or know where we can edit, pls help  me.
>
> BTW, I am using openser 1.2.2 version.
> Thanks in advance
> Tung
>
>
>
> ___
> Users mailing list
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> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>>>
> 
> 
> ___
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[OpenSIPS-Users] Mediaproxy ver 2.3.4 - Conntrack meets Music on Hold

[Stuart Marsden]

Hi,

I am testing a mediaproxy sever using 2.3.4

The tests I have done so far have worked, however there is one scenario 
where once the call has been setup a phone can generate Music on Hold

It does this by making a 2nd call (unrelated to the first at the SIP 
level) and passes what it thinks is  the IP:port of the 1st call far end 
(actually the mediaproxy) to the MoH server.

the MoH sever then sends audio to the mediaproxy.

However since the source of the audio is not the phone, conntrack is not 
forwarding on the audio to the other end.

Options as far as I can see are

1) remove the restriction on the source IP on who is allowed to forward 
rtp - from what I have ready this probably wont work
2) detect this special case and construct a temporary additional  "1 
way"  conntrack just for the MoH
3) detect this special case and change existing conntrack to point to 
the MoH sever, putting it back later

changes would be detected at the SIP level, or more likely conntrack 
call backs

any suggestions / help appreciated.

Stuart


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Re: [OpenSIPS-Users] OpenSIPS user meeting ?

[Bogdan-Andrei Iancu]

Hi,

bay2x1 wrote:
> That is a brilliant idea, specially for people like me who won't be able to
> travel  to America or Europe to attend the boot camp, because of time and
> financial constraints.  May I also suggest that some sort of online
> certification be offered for those who had taken the course.  I just hope
> this would be available on the soonest possible time.
>   
The whole certification test is anyhow done online (even from the 
bootcamps), so this is not an issue :)
> In regards to my former query, is there any available copy of the
> DVD-disc(the one distributed on the seminar) online for public download.  It
> will make studying opensips easier because I know that those configurations
> are working and that will serve as a reference of what I have configured.  I
> have been studying opensips for several months and its very difficult to
> find sample working configurations online that I can compare with my own
> configuration.  The only trusted reference I am using is the OpenSER book of
> Flavio E. Goncalves but some of the codes are obsolete or has been replaced.
The materials will be available as part of the e-Bootcamp course.

Regards,
Bogdan
>  
>
>
>
> Bogdan-Andrei Iancu wrote:
>   
>> I would love something like this :)
>>
>> Any other people willing to go for something like this ?
>>
>> Regards,
>> Bogdan
>>
>> Uwe Kastens wrote:
>> 
>>> Hello Bogdan,
>>>
>>> BTW: What about a opensips user meeting? I would prefer North of europe:)
>>>
>>>   
>>>   
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>> 
>
>   


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Re: [OpenSIPS-Users] database backend redundancy in opensips

[Bogdan-Andrei Iancu]

Hi Stefan,

right now you can do this directly from OpenSIPS (as application) - but 
maybe using a proxy DB in front.

But right now there is a new module under development to provide virtual 
DB URLs (for failover, balancing, parallel access to multiple DBs). It 
will be available in 1.6 in the next months.

Regards,
Bogdan

Stefan Tobé wrote:
> Hi, does anybody know whether opensips supports redundant database 
> backends?
>
> * if yes, how should that be configured?
> * If not supported, how could I achieve 'carrier grade' setup with
>   (multiple) opensips servers?
> * does opensips support opendbx or is it going to do so?
>
> thanks
> Stefan
>
>
>
>
> -- 
>
> Mvg
> Stefan Tobé
> 
>
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Re: [OpenSIPS-Users] opensips+dispatcher+asterisk problem

[Bogdan-Andrei Iancu]

Hi Ram,

I found your email on the Asterisk mailing list also ;)

So, to answer here also: do you get any reply back from Asterisk ?

Regards,
Bogdan

ram wrote:
> Hi all
>
> After a long iam back to forum
>  
> back to my own topic and several readings done on this forum
> how people doing same kind of setup what iam trying to achive
>
> so here i have done some good developements
>
> for testing iam doing all in one Server
>
> Step1 :
>
> Installed in Fresh BOX with Debian
>
> Asterisk and A2B working Fine
>
>
> Step2 : registered with SIP account iam able to make calls successfully
>
> Step3 :
>
> installed Opensips
>
> Made Subscribers to view from A2b Database
>
> Step4 : changed Asterisk port from 5060 to 5062
>
> Step5 : Opensip config made changes to register users with Opensips
> and when they dial 001X call send to Asterisk box
>
>
> route[3]{
>
> if (uri =~ "sip:001[0...@*"){
> log(1, "Forwarding to Asterisk \n");
> rewritehostport("A2b-asterisk-IP:5062");
> route(1);
> exit;
> }
>
> Works Fine, No problems as of now
>
> But to go in advance, i want to use Number of * boxes to achive more Load
>
> Step5 : added Dispatcher Module in the Opensips
>
> loadmodule "dispatcher.so"
> .
> .
> .
> modparam("dispatcher","list_file","/usr/local/etc/opensips/dispatcher.cfg")
> .
> .
> .
> .
> changed route to use dispatcher
>
> route[3]{
>
> if (uri =~ "sip:001[0...@*"){
> log(1, "Forwarding to Asterisk \n");
> ds_select_dst("2","4");
> forward();
> route(1);
> exit;
> }
>
>
> My dispatcher Config Looks like below
>
> dispatcher.cfg
> 2 sip:a2b-asterisk-ip:5062
> 2 sip:a2b-asterisk-ip2:5062
>
> I have restarted Opensips
>
> when i dial 0017XX number the call send Opensips to Asterisk
>
>
>
> Jun 30 01:12:28 opensips[25868]: Forwarding to Asterisk
> Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: set [2]
> Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: alg 
> hash [1]
> Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: 
> selected [4-2/1] 
> Jun 30 01:12:28 freeswitch opensips[25868]: DBG:core:mk_proxy: doing 
> DNS lookup...
> Jun 30 01:12:28 freeswitch opensips[25868]: DBG:core:forward_request: 
> sending:#012INVITE sip:0017x...@opensips-ip:5060 
> SIP/2.0#015#012Record-Route: #015#012Via: 
> SIP/2.0/UDP opensips-ip;branch=z9hG4bK28178282572929210914#015#012Via: 
> SIP/2.0/UDP 
> ip-phone-ip:5060;received=ip-phone-ip;branch=z9hG4bK28178282572929210914;rport=5060#015#012From:
>  
> 4720779942 ;tag=1966722825#015#012To: 
> 0017325824631 #015#012Call-ID: 
> 32167199575863-11502744529...@ip-phoneip#015#012cseq: 2 
> INVITE#015#012Contact: 
> #015#012Proxy-Authorization: Digest 
> username="4720779942", realm="asterisk", nonce="79ee65ba", 
> uri="sip:0017xxx...@opensips-ip:5060", 
> response="3e182f165a5663d0b145d6b55d34e94b", 
> algorithm=MD5#015#012Max-Forwards: 69#015#012Supported: 
> replaces#015#012User-Agent: Voip Phone 1.0#015#012Allow: INVITE, ACK, 
> OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE, PRACK, 
> UPDATE#015#012Content-Type: application/sdp#015#012Content-Length: 
> 319#015#012#015#012v=0#015#012o=4720779942 17025328 32005127 IN IP4 
> 202.63.111.2#015#012s=A conversation#015#012c=IN IP4 
> ip-phone-ip#015#012t=0 0#015#012m=audio 10028 RTP/AVP 18 4 8 0 9 
> 101#015#012a=rtpmap:18 G729/8000#015#012a=rtpmap:4 
> G723/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:0 
> PCMU/8000#015#012a=rtpmap:9 G722/16000#015#012a=rtpmap:101 
> telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=sendrecv#015#012.
> opensips[25868]: DBG:core:forward_request: orig. len=1087, 
> new_len=1220, proto=1
>
>
>
> when i ngrep
> 
>
>
> U 2009/06/30 01:59:20.770599 ipphone:5060 -> asterisk-a2b-ip:5060
> INVITE sip:0017x...@asterisk-a2b-ip:5060 SIP/2.0.
> Via: SIP/2.0/UDP ipphone:5060;branch=z9hG4bK2932733762726732719;rport.
> From: 4720779942 ;tag=3037030266.
> To: 0017 .
> Call-ID: 14399316162240-7371067914...@ipphone.
> CSeq: 2 INVITE.
> Contact: .
> Proxy-Authorization: Digest username="4720779942", realm="asterisk", 
> nonce="07ba8624", uri="sip:0017x...@asterisk-a2b-ip:5060", 
> response="5dbe9b2937d0bc3f6e8d25052fff0b6a", algorithm=MD5.
> Max-Forwards: 70.
> Supported: replaces.
> User-Agent: Voip Phone 1.0.
> Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, 
> SUBSCRIBE, PRACK, UPDATE.
> Content-Type: application/sdp.
> Content-Length: 319.
> .
> v=0.
> o=4720779942 69102627 18481147 IN IP4 ipphone.
> s=A conversation.
> c=IN IP4 ipphone.
> t=0 0.
> m=audio 10034 RTP/AVP 18 4 8 0 9 101.
> a=rtpmap:18 G729/8000.
> a=rtpmap:4 G723/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:9 G722/16000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=sendrecv.
>
>
> U 2009/06/30 01:59:20.774528 asterisk-a2b-ip:5060 -> ipphone:5060
> SIP/2.0 100 Giving a try.
> Via: SIP/2.0/UDP 
> ipphone:5060;branch=z9hG4bK2932733762726732719;rport=5060.
> From: 4720779942 ;tag=3037030266.
> To: 0017 .
> C

Re: [OpenSIPS-Users] Registration and Loose-Route

[Bogdan-Andrei Iancu]

Hi Nathaniel,

OpenSIPs gives you the 403 as it does not recognize itself in the Route 
hdr of the REGISTER. By adding the entry in the domain table, OpenSIPS 
will recognize its own SIP domains and it will validate the request.

about the 401 - this is the normal step during the authentication 
process. First the UAC sends a request with no credentials, the proxy 
answers with a 401 asking for auth; then the UAC should send a new 
request containing the credentials.

Regards,
Bogdan

Nathaniel L Keeling wrote:
> If there is no entry in the domain table, the it will error in the 
> loose_route() function and the error message that I get is "403 
> Preload Route denied". When I add an entry to the domain table, it 
> passes the loose_route() function and then error while authenticating. 
> I have placed an xlog statement within the register block of the 
> config file and right before the loose_route() function block is 
> executed. I have included my config file.
>
> thanks
>
> Nathaniel
>
> Eduardo Panciera wrote:
>> Are you sure that the message are been processed by a register block 
>> of your configuration? can you attach your configuration file? you 
>> can use log function in the differents blocks of your configuration , 
>> in order to clarify your debug.
>>  
>> best regards.
>> Pancho.
>>
>> On Mon, Jun 29, 2009 at 9:06 PM, Nathaniel L Keeling 
>> mailto:keel...@akan-tech.com>> wrote:
>>
>> I am new and need an explanation. I have installed opensips 1.5 with
>> database support. I am trying to authenticate via the subscriber's
>> table. Utilizing the sample config file and uncommenting the 
>> areas to
>> allow authentication via database, I try to register a sip device. I
>> have added a user using opensipsctl. When the registration requests
>> comes in, it dies in the loose_route() function with the error "403
>> Preload Route Denied". According to the documentation on the
>> loose_route() function, if there is no to-tag and there is only on
>> route
>> header indicating the localproxy, the function should return 
>> false. It
>> is returning true. I then added the sip domain to the domain 
>> table and
>> the error changes to "401Unauthorized". Please explain. I am 
>> including
>> the SIP message and the debug output.
>>
>> Jun 29 01:15:03 [15473] DBG:core:parse_msg: SIP Request:
>> Jun 29 01:15:03 [15473] DBG:core:parse_msg:  method:  
>> Jun 29 01:15:03 [15473] DBG:core:parse_msg:  uri:
>> > >
>> Jun 29 01:15:03 [15473] DBG:core:parse_msg:  version: 
>> Jun 29 01:15:03 [15473] DBG:core:parse_headers: flags=2
>> Jun 29 01:15:03 [15473] DBG:core:parse_via_param: found param type
>> 232,
>>  = ; state=6
>> Jun 29 01:15:03 [15473] DBG:core:parse_via_param: found param type
>> 235,
>>  = ; state=17
>> Jun 29 01:15:03 [15473] DBG:core:parse_via: end of header reached,
>> state=5
>> Jun 29 01:15:03 [15473] DBG:core:parse_headers: via found, flags=2
>> Jun 29 01:15:03 [15473] DBG:core:parse_headers: this is the first 
>> via
>> Jun 29 01:15:03 [15473] DBG:core:receive_msg: After parse_msg...
>> Jun 29 01:15:03 [15473] DBG:core:receive_msg: preparing to run 
>> routing
>> scripts...
>> Jun 29 01:15:03 [15473] DBG:core:parse_headers: flags=100
>> Jun 29 01:15:03 [15473] DBG:core:parse_to: end of header reached,
>> state=10
>> Jun 29 01:15:03 [15473] DBG:core:parse_to: display={},
>> ruri={sip:3124530...@kwesi.chicagosip1.akan.us.com
>> }
>> Jun 29 01:15:03 [15473] DBG:core:get_hdr_field:  [48];
>> uri=[sip:3124530...@kwesi.chicagosip1.akan.us.com
>> ]
>> Jun 29 01:15:03 [15473] DBG:core:get_hdr_field: to body
>> [> >
>> ]
>> Jun 29 01:15:03 [15473] DBG:core:get_hdr_field: cseq : <6493>
>> 
>> Jun 29 01:15:03 [15473] DBG:maxfwd:is_maxfwd_present: value = 70
>> Starting to process request
>> Jun 29 01:15:03 [15473] DBG:uri:has_totag: no totag
>> we are about to check for cancel
>> Jun 29 01:15:03 [15473] DBG:core:parse_headers: flags=78
>> Jun 29 01:15:03 [15473] DBG:tm:t_lookup_request: start searching:
>> hash=15692, isACK=0
>> Jun 29 01:15:03 [15473] DBG:tm:matching_3261: RFC3261 transaction
>> matching failed
>> Jun 29 01:15:03 [15473] DBG:tm:t_lookup_request: no transaction 
>> found
>> we are about to check registration and multidomain
>> we are about to check for loose route
>> Jun 29 01:15:03 [15473] DBG:core:parse_headers: flags=200
>> Jun 29 01:15:03 [15473] DBG:rr:is_preloaded: is_preloaded: Yes
>> Jun 29 01:15:03 [15473] DBG:core:grep_sock_info: checking if 
>> host==us:
>> 29

Re: [OpenSIPS-Users] SOLVED: Re: OpenSIPS 1.5.1 with Postgres

[Bogdan-Andrei Iancu]

Hi Gordon,

historical reasons - the module was named "postgres" and changing the 
names of the module is not desirable. And the convention is to build the 
DB URLs with the name of the module (to specify the DB type to use)

So, more or less we are stuck with POSTGRES :)

Regards,
Bogdan

Gordon Ross wrote:
> BTW, why the desire to use "POSTGRES" rather than "PGSQL" ?
>
> GTG
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[OpenSIPS-Users] database backend redundancy in opensips

[Stefan Tobé]

Hi, does anybody know whether opensips supports redundant database backends?

   - if yes, how should that be configured?
   - If not supported, how could I achieve 'carrier grade' setup with
   (multiple) opensips servers?
   - does opensips support opendbx or is it going to do so?

thanks
Stefan




-- 

Mvg
Stefan Tobé
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Re: [OpenSIPS-Users] [Open-sips]: Problem in message body of Notify

[Anca Vamanu]

Hi Shefali,

I checked also and you are right. I have fixed this in trunk, 1.5.x and 
1.4.x svn branch.

Thanks and regards,
Anca


Shefali Dutta wrote:
>
> Hi,
>
> Open-sips is send Notify with MIME message body which seems to be 
> incorrect as per RFC.
>
> According to RFC 2046:
>
> “The boundary delimiter MUST occur at the beginning of a line, i.e.,
>following a CRLF, and the initial CRLF is considered to be attached
>to the boundary delimiter line rather than part of the preceding
>part.  The boundary may be followed by zero or more characters of
>linear whitespace. *It is then terminated by either another CRLF and*
> *   the header fields for the next part, or by two CRLFs, in which case*
> *   there are no header fields for the next part.*  If no Content-Type
>field is present it is assumed to be "message/rfc822" in a
>"multipart/digest" and "text/plain" otherwise.”
>  
> Open-sips is sending an extra CRLF after the boundary between different 
> message bodies as a result, MIME headers are also assumed to be part of the 
> body by the parser.
> Here is the message dump of Notify:
>
> -
>
> NOTIFY sip:pri...@172.16.107.44;transport=udp SIP/2.0
>
> Via: SIP/2.0/UDP 172.16.107.83;branch=z9hG4bK4bc1.29ea1ec.0
>
> To: sip:pri...@172.16.107.83;tag=ICF_808806399-18528
>
> From: sip:gro...@172.16.107.83;tag=10.8289.1245842000.392
>
> CSeq: 2 NOTIFY
>
> Call-ID: 808806272-18...@172.16.107.44
>
> Content-Length: 1178
>
> User-Agent: OpenSIPS (1.4.4-notls (i386/linux))
>
> Max-Forwards: 70
>
> Event: presence
>
> Contact: 
>
> Subscription-State: active;expires=7196
>
> Require: eventlist
>
> Content-Type: multipart/related;type="application/rlmi+xml";start= 
> "<1245842004.sip:gro...@172.16.107.83.1189641421>";boundary="4wUh1TIgBYw5MoUErujY6lh5"
>
> --4wUh1TIgBYw5MoUErujY6lh5
>
> Content-Transfer-Encoding: binary
>
> Content-ID: <1245842004.sip:gro...@172.16.107.83.1189641421>
>
> Content-Type: application/rlmi+xml;charset="UTF-8r"
>
> 
>
>  xmlns="urn:ietf:params:xml:ns:rlmi" version="2" fullState="false">
>
> 
>
>  cid="1245842004.sip:pran...@172.16.107.83.596516649"/>
>
> 
>
> 
>
> *--4wUh1TIgBYw5MoUErujY6lh5*
>
> * *
>
> *Content-Transfer-Encoding: binary*
>
> Content-ID: <1245842004.sip:pran...@172.16.107.83.596516649>
>
> Content-Type: application/pidf+xml
>
> 
>
>  xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" 
> xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" 
> xmlns:c="urn:ietf:params:xml:ns:pidf:cipid" 
> entity="sip:pran...@172.16.107.83">
>
> 
>
> 
>
> open
>
> 
>
> 
>
> 
>
> 
>
> 
>
> 
>
> Available
>
> 
>
> 
>
> --4wUh1TIgBYw5MoUErujY6lh5--
>
> 
>
>  
>
> Is there any patch available for this problem in opensips.
>
> Rgds,
>
> Shefali
>
>
> 
> "DISCLAIMER: This message is proprietary to Aricent and is intended 
> solely for the use of the individual to whom it is addressed. It may 
> contain privileged or confidential information and should not be 
> circulated or used for any purpose other than for what it is intended. 
> If you have received this message in error,please notify the 
> originator immediately. If you are not the intended recipient, you are 
> notified that you are strictly prohibited from using, copying, 
> altering, or disclosing the contents of this message. Aricent accepts 
> no responsibility for loss or damage arising from the use of the 
> information transmitted by this email including damage from virus."
> 
>
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[OpenSIPS-Users] opensips+dispatcher+asterisk problem

[ram]

Hi all

After a long iam back to forum

back to my own topic and several readings done on this forum
how people doing same kind of setup what iam trying to achive

so here i have done some good developements

for testing iam doing all in one Server

Step1 :

Installed in Fresh BOX with Debian

Asterisk and A2B working Fine


Step2 : registered with SIP account iam able to make calls successfully

Step3 :

installed Opensips

Made Subscribers to view from A2b Database

Step4 : changed Asterisk port from 5060 to 5062

Step5 : Opensip config made changes to register users with Opensips
and when they dial 001X call send to Asterisk box


route[3]{

if (uri =~ "sip:001[0...@*"){
log(1, "Forwarding to Asterisk \n");
rewritehostport("A2b-asterisk-IP:5062");
route(1);
exit;
}

Works Fine, No problems as of now

But to go in advance, i want to use Number of * boxes to achive more Load

Step5 : added Dispatcher Module in the Opensips

loadmodule "dispatcher.so"
.
.
.
modparam("dispatcher","list_file","/usr/local/etc/opensips/dispatcher.cfg")
.
.
.
.
changed route to use dispatcher

route[3]{

if (uri =~ "sip:001[0...@*"){
log(1, "Forwarding to Asterisk \n");
ds_select_dst("2","4");
forward();
route(1);
exit;
}


My dispatcher Config Looks like below

dispatcher.cfg
2 sip:a2b-asterisk-ip:5062
2 sip:a2b-asterisk-ip2:5062

I have restarted Opensips

when i dial 0017XX number the call send Opensips to Asterisk



Jun 30 01:12:28 opensips[25868]: Forwarding to Asterisk
Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: set [2]
Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: alg hash [1]
Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: selected
[4-2/1] 
Jun 30 01:12:28 freeswitch opensips[25868]: DBG:core:mk_proxy: doing DNS
lookup...
Jun 30 01:12:28 freeswitch opensips[25868]: DBG:core:forward_request:
sending:#012INVITE sip:0017x...@opensips-ip:5060
SIP/2.0#015#012Record-Route: #015#012Via: SIP/2.0/UDP
opensips-ip;branch=z9hG4bK28178282572929210914#015#012Via: SIP/2.0/UDP
ip-phone-ip:5060;received=ip-phone-ip;branch=z9hG4bK28178282572929210914;rport=5060#015#012From:
4720779942 ;tag=1966722825#015#012To:
0017325824631 #015#012Call-ID:
32167199575863-11502744529...@ip-phoneip#015#012cseq: 2
INVITE#015#012Contact:
#015#012Proxy-Authorization:
Digest username="4720779942", realm="asterisk", nonce="79ee65ba",
uri="sip:0017xxx...@opensips-ip:5060",
response="3e182f165a5663d0b145d6b55d34e94b",
algorithm=MD5#015#012Max-Forwards: 69#015#012Supported:
replaces#015#012User-Agent: Voip Phone 1.0#015#012Allow: INVITE, ACK,
OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE, PRACK,
UPDATE#015#012Content-Type: application/sdp#015#012Content-Length:
319#015#012#015#012v=0#015#012o=4720779942 17025328 32005127 IN IP4
202.63.111.2#015#012s=A conversation#015#012c=IN IP4 ip-phone-ip#015#012t=0
0#015#012m=audio 10028 RTP/AVP 18 4 8 0 9 101#015#012a=rtpmap:18
G729/8000#015#012a=rtpmap:4 G723/8000#015#012a=rtpmap:8
PCMA/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:9
G722/16000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101
0-15#015#012a=sendrecv#015#012.
opensips[25868]: DBG:core:forward_request: orig. len=1087, new_len=1220,
proto=1



when i ngrep



U 2009/06/30 01:59:20.770599 ipphone:5060 -> asterisk-a2b-ip:5060
INVITE sip:0017x...@asterisk-a2b-ip:5060 SIP/2.0.
Via: SIP/2.0/UDP ipphone:5060;branch=z9hG4bK2932733762726732719;rport.
From: 4720779942 ;tag=3037030266.
To: 0017 .
Call-ID: 14399316162240-7371067914...@ipphone.
CSeq: 2 INVITE.
Contact: .
Proxy-Authorization: Digest username="4720779942", realm="asterisk",
nonce="07ba8624", uri="sip:0017x...@asterisk-a2b-ip:5060",
response="5dbe9b2937d0bc3f6e8d25052fff0b6a", algorithm=MD5.
Max-Forwards: 70.
Supported: replaces.
User-Agent: Voip Phone 1.0.
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE,
PRACK, UPDATE.
Content-Type: application/sdp.
Content-Length: 319.
.
v=0.
o=4720779942 69102627 18481147 IN IP4 ipphone.
s=A conversation.
c=IN IP4 ipphone.
t=0 0.
m=audio 10034 RTP/AVP 18 4 8 0 9 101.
a=rtpmap:18 G729/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:9 G722/16000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.


U 2009/06/30 01:59:20.774528 asterisk-a2b-ip:5060 -> ipphone:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP ipphone:5060;branch=z9hG4bK2932733762726732719;rport=5060.
From: 4720779942 ;tag=3037030266.
To: 0017 .
Call-ID: 14399316162240-7371067914...@ipphone.
CSeq: 2 INVITE.
Server: OpenSIPS (1.5.1-notls (i386/linux)).
Content-Length: 0.
.


U 2009/06/30 01:59:21.650498 asterisk-a2b-ip:5060 -> ipphone:5060
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP
ipphone:5060;received=ipphone;branch=z9hG4bK1984515716453028636;rport=5060.
From: 4720779942 ;tag=3037030266.
To: 0017 ;tag=as0cb075c5.
Call-ID: 14399316162240-7371067914...@ipphone.
CSeq: 1 INVITE.
User-Agent: Asterisk PBX.
Al

Re: [OpenSIPS-Users] 1.6 and mysql => crash Re: DIALOG not deleted on BYE

[Bogdan-Andrei Iancu]

Hi Uwe,

I see the core was not generated, so no bt :(can you reproduce the 
crash? can you get a core file and a bt ?

Thanks and regards,
Bogdan

Uwe Kastens wrote:
> Bogdan,
>
> Sorry for bothering again. I tried the latest trunk from svn and
> opensips is dying after accessing the mysql db.
>
> I will attach the trace.
>
> BR
>
> Uwe
>
>
>
> Bogdan-Andrei Iancu schrieb:
>   
>> OK - with the fix from SVN you should be able to call loose_route() as
>> many times you want without any risk - just let me know if it works as
>> expected.
>>
>> Regards,
>> Bogdan
>>
>> Uwe Kastens wrote:
>> 
>>> Hi Bogdan,
>>>
>>> Again, thanks a lot for your help.
>>>
>>> The loose_route() seems to cause the problem, but somehow its needed to
>>> pass byes correctly to the UA. So I need to work a little on my skript.
>>>
>>> I will try the 1.6 ASAP and let you know the result.
>>>
>>> BR
>>>
>>> Uwe
>>>
>>>
>>>
>>> Bogdan-Andrei Iancu schrieb:
>>>  
>>>   
 If you could test, a fix is available on 1.6 (trunk) version - if ok, I
 will do the backport.

 Regards,
 Bogdan

 Bogdan-Andrei Iancu wrote:

 
> Hi Uwe,
>
> Thanks for the traces. Looking at the opensips logs, I say you do
> loose_route() twice for the ACK which looks twice for the dialog and
> increase the ref twice for the dialogthis is why the ref never
> gets back to 0 to allow the dialog to be destroyed..
>
> Could you confirm this for me ?
>
> even if it's a script error , the dialog module should cope with it..I
> will look for a fix.
>
> Thanks and regards,
> Bogdan
>
> Bogdan-Andrei Iancu wrote:
>  
>  
>   
>> Hi Uwe,
>>
>>
>> Uwe Kastens wrote:
>> 
>> 
>>> Hi again,
>>>
>>> So I think it might be a bug. One direction (UA to PSTN) works
>>> everytime
>>> perfectly. It doesn't matter on which side the BYE is sent. If I try
>>> the
>>> other direction, the dialog will not be removed. Again it won't
>>> matter
>>> on which side the BYE is sent - the dialog will stay active.
>>>   
>>>   
>> yes, it sounds like.
>> 
>> 
>>> Unfort I was not able to find out what the states and the events
>>> means.
>>>   
>>>   
>> You can find the meaning of each state in: modules/dialog/dlg_hash.h
>>
>>
>> 
>> 
>>> So its not easy to debug further.
>>>
>>> Working direction:
>>> DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 1 to
>>> state 2, due event 2
>>> DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 2 to
>>> state 3, due event 3
>>> DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 3 to
>>> state 4, due event 6
>>> DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to
>>> state 4, due event 6
>>> DBG:dialog:next_state_dlg: dialog 0xd7a30870 changed from state 4 to
>>> state 4, due event 1
>>>
>>> Not Working
>>> DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 1 to
>>> state 2, due event 2
>>> DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to
>>> state 2, due event 2
>>> DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 2 to
>>> state 3, due event 3
>>> DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 3 to
>>> state 5, due event 7
>>> DBG:dialog:next_state_dlg: dialog 0xd7a2c6e0 changed from state 5 to
>>> state 5, due event 1
>>>
>>> Anyone could help please?
>>>   
>>>   
>> I can try : )
>>
>> could you (privately if needed) please send me the the full logs for
>> the entire call (debug=6) - for the non working part.
>>
>> Thanks and regards,
>> Bogdan
>> 
>> 
>>> BR
>>>
>>> Uwe
>>>
>>>
>>> Uwe Kastens schrieb:
>>> 
>>>   
 Hello again,

 I think the dialog is destroyed, if no reference is left. And so I
 asume
  the dialog is missing the ACK for the BYE. Or do I need to unref it
 manually  via reply_route? I will attach the log.

 dialog::  hash=440:1838775488
 state:: 5
 user_flags:: 0
 timestart:: 1246005835
 timeout:: 0
 callid:: 240f6fed145ac8251915f50d3d54b...@10.20.138.105
 from_uri:: sip:9904...@10.20.138.105:5100
 from_tag:: as619609ab
 caller_contact:: sip:9904...@10.20.138.105:5100
 caller_cseq:: 102
 caller_route_set::
 caller_bind_addr:: udp:10.20.138.125:5100
 to_u