Re: [OpenSIPS-Users] dlg_validate_dialog errors in loose_route

2011-04-12 Thread Jeff Pyle
This thing is picky… The validation failed with the following values in the log 
entry:

req=[sip:;lr;r2=on;ftag=gK0706089a;did=e17.bfb88ba7],
dlg=[sip:;r2=on;lr;ftag=gK0706089a;did=e17.bfb88ba7]

Not quite sure what to say about this one.


- Jeff

From: Jeff Pyle mailto:jp...@fidelityvoice.com>>
Reply-To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Date: Mon, 11 Apr 2011 10:15:11 -0400
To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Subject: Re: [OpenSIPS-Users] dlg_validate_dialog errors on BYE and ACK in 
loose_route

Indeed!  I hadn't noticed the difference in the dlg vs req sections of the 
debug message.

I shall try exactly as you suggest.  Thanks.


- Jeff

From: Vlad Paiu mailto:vladp...@opensips.org>>
Reply-To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Date: Mon, 11 Apr 2011 06:13:32 -0400
To: "users@lists.opensips.org" 
mailto:users@lists.opensips.org>>
Subject: Re: [OpenSIPS-Users] dlg_validate_dialog errors on BYE and ACK in 
loose_route

Hello Jeff,

It's not that the dialog is already torn down by the time the BYE gets to that 
particular portion of script.
It's complaining about the fact that at the dialog creation, the route had the 
"lr=on" parameter, and at BYE time, the parameter has been stripped and set
just to "lr".

This should not happen in a regular scenario, as other SIP entities should not 
explicitly modify the content of Route headers recorded at Invite time.
You could try to check the  enable_full_lr ( [1] ) parameter, and onlyattach 
"lr" instead of "lr=on", and see if this fixes the behavior.

[1] http://www.opensips.org/html/docs/modules/devel/rr.html#id250149

Regards,

--
Vlad Paiu
OpenSIPS Developer


On 04/09/2011 10:32 PM, Jeff Pyle wrote:
Hello,

I'm trying a new chunklet of config in loose_route().  I believe this came from 
a documentation page somewhere, so I don't pretend to take credit for the 
concept.

# Clean up dialog data if necessary
if ($DLG_status!=NULL && !validate_dialog()) {
xlog("L_INFO", "Unable to validate dialog, 
fixing - $hdr(CSeq) on $ci\n");
fix_route_dialog();
}

I send the first batch of pseudo-production traffic through this config today 
and I ended up with a metric ton of:
  ERROR:dialog:dlg_validate_dialog: Check failed for routenumber 0. 
req=[sip:;lr=on;ftag=23ddd50-0-13c4-1f179b-78884527-1f179b;did=45b.b972314]

And, the xlogs from the above line:
  Unable to validate dialog, fixing - 3 BYE on 

It showed up on a number of BYEs and ACKs.  My dialog module config is as 
follows:

# - dialog params -
modparam("dialog", "dlg_flag", 4)
modparam("dialog", "bye_on_timeout_flag", 4)
modparam("dialog", "default_timeout", 21600)
modparam("dialog|sst", "timeout_avp", "$avp(s:dialog_timeout)")
modparam("dialog", "dlg_extra_hdrs", "Hint: dialog expired\r\n")
modparam("dialog", "dlg_match_mode", 1)  # 0 = DID_ONLY; 1 = DID first, SIP 
second; 2 = SIP only
modparam("dialog", "db_url", "mysql://osuer:ospass@dbcluster/osdb")
modparam("dialog", "table_name", "dialog_1")
modparam("dialog", "db_mode", 1)  # 1 = Realtime to DB; 2 = delayed; 3 = on 
shutdown
modparam("dialog", "profiles_with_value", "711calls; 729calls; calls")

create_dialog() runs on all initial INVITEs.

Is this type of behavior anything close to normal?  What might cause it?  My 
immediate fix is to simply comment out this portion of the config, but I'm 
afraid I might be masking a larger issue.  Any suggestions would be great.

Just had a thought as I was about to send this email.  Perhaps this is because 
by the time the BYE gets to this portion of the script, the dialog is already 
torn down?  Could it be that simple, just add "if !(is_method("BYE") || 
is_method("ACK"))" to the top line of this config portion?


- Jeff



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Re: [OpenSIPS-Users] codec_exists and its cousins on 1.6.4

2011-04-12 Thread Jeff Pyle
Never mind!  Turned out to be an extra \r\n in an append_rf, set once on the 
variable that the function used, and a second time on the function itself.  As 
you were…

I love it when it's easy.

Strange, though, the same config on 1.6.2 didn't cause the carriers to barf on 
it.


- Jeff

From: Jeff Pyle mailto:jp...@fidelityvoice.com>>
Reply-To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Date: Tue, 12 Apr 2011 17:12:37 -0400
To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Subject: [OpenSIPS-Users] codec_exists and its cousins on 1.6.4

Hey list,

Has anyone heard of a problem where codec_exists, codec_delete_except_re, 
codec_move_up, or codec_delete causes an extra blank line to appear between the 
headers and the SDP in 1.6.4?  I updated from 1.6.2 and I have this now.  
Upstream gateways bounce the invites with 400 Bad Request.

I haven't been able to determine exactly which function causes the problem.  I 
thought I'd ask before I rip apart an otherwise functional config.


- Jeff

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[OpenSIPS-Users] codec_exists and its cousins on 1.6.4

2011-04-12 Thread Jeff Pyle
Hey list,

Has anyone heard of a problem where codec_exists, codec_delete_except_re, 
codec_move_up, or codec_delete causes an extra blank line to appear between the 
headers and the SDP in 1.6.4?  I updated from 1.6.2 and I have this now.  
Upstream gateways bounce the invites with 400 Bad Request.

I haven't been able to determine exactly which function causes the problem.  I 
thought I'd ask before I rip apart an otherwise functional config.


- Jeff

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Re: [OpenSIPS-Users] MediaProxy and redirected traffic question

2011-04-12 Thread Saúl Ibarra Corretgé

Hi,

On 12/4/11 3:11 PM, Stylianos Papanastasiou wrote:

Hi Saúl,

and thanks for taking the time to reply - I have since made some
progress into understanding the terminal behaviour with respect to
what I am trying to achieve but have not got to a solution to my
problem of having permanently redirected traffic.

Here is what happened in my scenario:

After inserting the iptables rule above all packets coming from
terminal 10.0.0.2 were redirected to my machine on port 4. so
assume that communications between media proxy and the terminal are as
so:

terminalmedia proxy  iptables redirection
10.0.0.2:9078 >10.0.0.1:50002 >  10.0.0.1:4

Now, as I intercept all traffic from 10.0.0.3 (that is the other
terminal), I add some effect and repackage it with gstreamer, and sent
it to 10.0.0.2
Since the port 50002 is bound in media proxy, however, I send traffic
from a different local port, say 30006. So we have

terminal media proxy
10.0.0.2:9078< 10.0.0.1:30006

My terminal clients are linphone clients and they send video traffic
back to the new port (not 50002 anymore but 30006 instead) - so we
have:

terminal
10.0.0.2:9078-->  10.0.0.1:30006



At this point, the client switched the port because it is receiving 
traffic from another place, right? Because no re-INVITE happened, 
MediaProxy is not aware of this.



I cannot understand, hard though I might try, why the rule
iptables -t nat -A PREROUTING -i eth0  -p udp -s 10.0.0.2 --sport 9078
-j REDIRECT --to-port 4
does not keep working in that case and does not keep redirecting
traffic to port 4. I see traffic on port 30006 instead (and I can,
of course, decode and manipulate it).

If anyone has an idea why that happens please consider dropping a hint.

Another way to solve my problems would be to alter mediaproxy to place
conntrack rules that forward traffic to a local port instead of what
is on the other side in the session description. Again if someone
could point to the right place in the source code, I would be very
grateful.



I think this should be the way to go in your very specific situation. 
Mangling iptables rules will definitely affect how MediaProxy works or 
actually does not work.


I guess you'll want to hack MediaProxy to create conntrack rules 
differently. Have a look at the check_create_conntrack function on 
mediaproxy/mediacontrol.py file, conntrack rules are created there as 
soon as 1 RTP packet is received from both sides.



Regards,

--
Saúl Ibarra Corretgé
AG Projects

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[OpenSIPS-Users] groupID avp with do_routing

2011-04-12 Thread Richard Revels
Hope this saves someone time at some point.  The type of value stored in the 
groupID avp (if used) when calling do_routing() must be an int.  There is no 
attempt to convert in the function.  I store everything in usr_preferences as a 
string and then use transforms where needed in the config script.  Took me a 
day to realize I needed one for this.

Richard
 
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Re: [OpenSIPS-Users] uac_replace_from auto mode with to_tag ignores display name

2011-04-12 Thread Jeff Pyle
Razvan,

On the initial INVITE the first parameter is not set to "".  I run the function 
as follows:

uac_replace_from("$var(id_namereplace)", 
"sip:$avp(s:id_num)@$Ri");

On the initial INVITE, both the display name and the URI are updated correctly, 
that is, with the values put into the function.  The problem happens in 
loose_route when the URI is updated but the display name is not.

To get around the problem of the original display name leaking through loose 
route, I have added the following functions to neutralize it completely.  No 
display name on subsequent requests is better than the wrong name:

if (is_direction("downstream")) uac_replace_from("","");
if (is_direction("upstream")) uac_replace_to("","");

This does effectively remove the display name without touching the URI.  But 
shouldn't the display name be updated automatically (in auto mode) just as the 
URI is?


- Jeff


From: Razvan Crainea 
mailto:razvancrai...@opensips.org>>
Reply-To: OpenSIPS users mailling list 
mailto:users@lists.opensips.org>>
Date: Tue, 12 Apr 2011 03:42:44 -0400
To: "users@lists.opensips.org" 
mailto:users@lists.opensips.org>>
Subject: Re: [OpenSIPS-Users] uac_replace_from auto mode with to_tag ignores 
display name

Hello, Jeff,

Please make sure that the first parameter is not set to "". As you can see 
here[1], if the first parameter of uac_replace_from method is "", it removes 
the Display name.

Regards,
Razvan

[1] http://www.opensips.org/html/docs/modules/devel/uac.html#id292701

On 04/11/2011 07:03 PM, Jeff Pyle wrote:
Hello,

I run uac_replace_from (auto mode) in a branch_route.  Subsequents requests for 
the same dialog (with to_tag, through loose_route) have the From URI updated 
correctly, but the display name isn't.  Is this a misconfiguration on my part, 
by design, or a bug?


- Jeff



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Re: [OpenSIPS-Users] OpenSIPS-CP - Call to undefined method MDB2_Error::setFetchMode()

2011-04-12 Thread duane . larson
I did create the admin user in the opc_admin_privileges table. It was a DB  
connection. In the db.inc.php I changed the


//database host
$config->db_host = "localhost";

to
//database host
$config->db_host = "127.0.0.1";


Can't believe I missed that. Thanks!!

On Apr 12, 2011 3:18am, Alex Ionescu  wrote:

Hi Duane,







Have you created the initial user in the opc_admin_privileges table ?




You have to insert an user - it is written in the INSTALL file.



Anyway, if you did, please check again if the connection details  
in ../opensips-cp/config/db.inc.php are the correct ones.



Unfortunatelly, MDB2 is not very "friendly" when reporting errors ... it  
may create confusion.







Let me know if you succeeded!







Regards,




Alex







Anton Zagorskiy wrote:






Hi Duane.







In most cases it means that it can't connect to DB.
















From: users-boun...@lists.opensips.org




[mailto:users-boun...@lists.opensips.org] On Behalf Of




duane.lar...@gmail.com




Sent: Tuesday, April 12, 2011 3:44 AM




To: OpenSIPS users mailling list




Subject: [OpenSIPS-Users] OpenSIPS-CP - Call to undefined method




MDB2_Error::setFetchMode()






I am getting this error with OpenSIPS-CP when I try to log in for the  
first




time










Fatal error: Call to undefined method MDB2_Error::setFetchMode() in




/var/www/opensips-cp/web/db_connect.php on line 31






I saw that someone else in the mailing list had the exact same error and  
it




was mentioned that the db.ini.php wasn't configured. As far as I know I




followed the installation instructions. I installed all the pear packages




and edited any files that were supposed to be edited. Not sure what's




causing this. I downloaded with svn.










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Re: [OpenSIPS-Users] MediaProxy and redirected traffic question

2011-04-12 Thread Stylianos Papanastasiou
Hi Saúl,

and thanks for taking the time to reply - I have since made some
progress into understanding the terminal behaviour with respect to
what I am trying to achieve but have not got to a solution to my
problem of having permanently redirected traffic.

Here is what happened in my scenario:

After inserting the iptables rule above all packets coming from
terminal 10.0.0.2 were redirected to my machine on port 4. so
assume that communications between media proxy and the terminal are as
so:

terminalmedia proxy  iptables redirection
10.0.0.2:9078 >   10.0.0.1:50002 > 10.0.0.1:4

Now, as I intercept all traffic from 10.0.0.3 (that is the other
terminal), I add some effect and repackage it with gstreamer, and sent
it to 10.0.0.2
Since the port 50002 is bound in media proxy, however, I send traffic
from a different local port, say 30006. So we have

terminal media proxy
10.0.0.2:9078 < 10.0.0.1:30006

My terminal clients are linphone clients and they send video traffic
back to the new port (not 50002 anymore but 30006 instead) - so we
have:

terminal
10.0.0.2:9078--> 10.0.0.1:30006

I cannot understand, hard though I might try, why the rule
iptables -t nat -A PREROUTING -i eth0  -p udp -s 10.0.0.2 --sport 9078
-j REDIRECT --to-port 4
does not keep working in that case and does not keep redirecting
traffic to port 4. I see traffic on port 30006 instead (and I can,
of course, decode and manipulate it).

If anyone has an idea why that happens please consider dropping a hint.

Another way to solve my problems would be to alter mediaproxy to place
conntrack rules that forward traffic to a local port instead of what
is on the other side in the session description. Again if someone
could point to the right place in the source code, I would be very
grateful.

Having said that, even having received this much help went beyond my
expectations.

Thanks again,
Stylianos


On 12 April 2011 09:21, Saúl Ibarra Corretgé  wrote:
> Hi,
>
> On 5/4/11 9:54 PM, Stylianos Papanastasiou wrote:
>>
>> Hi everyone,
>>
>> I am running opensips with mediaproxy and so the media traffic from my
>> two SIP terminals runs through an intermediate machine. Now, I want to
>> do some media traffic manipulation so I redirect the incoming traffic to
>> some port on the intermediate machine using iptables. Assuming my
>> terminals are 10.0.0.2 and 10.0.0.3 and the media-proxy is 10.0.0.1 -
>> and the video traffic comes from port 9078 from both terminals I give:
>>
>> iptables -t nat -A PREROUTING -i eth0  -p udp -s 10.0.0.2 --sport 9078
>> -j REDIRECT --to-port 4
>> iptables -t nat -A PREROUTING -i eth0  -p udp -s 10.0.0.3 --sport 9078
>> -j REDIRECT --to-port 40001
>>
>
> MediaProxy does insert contrack rules to direct the traffic, as you know,
> and the fact that you are altering the routing with iptables may have
> serious implications on what happens from that point on. MediaProxy told the
> kernel to redirect packets coming from 10.0.0.2 and port X to 10.0.0.3 port
> Y, but you introduced another routing rule I don't expect packets to 'fork'.
>
>
> --
> Saúl Ibarra Corretgé
> AG Projects
>
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[OpenSIPS-Users] [REMINDER] OpenSIPS IRC meeting, 13th of April, 17:00 PM CET

2011-04-12 Thread Bogdan-Andrei Iancu

Topic:

http://www.opensips.org/Resources/IRCmeeting20110413

Regards,
Bogdan

--
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OpenSIPS eBootcamp - 2nd of May 2011
OpenSIPS solutions and "know-how"


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[OpenSIPS-Users] OpenSIPS EBootcamp - new session

2011-04-12 Thread Bogdan-Andrei Iancu

Hi all,

Following the user demand on the OpenSIPS eBootcamp training, a new 
session was scheduled for

*2nd of May 2011*

The eBootcamp session targets OpenSIPS 1.6.4 with the following content:
SIP introduction
OpenSIPS introduction
SQL support (auth, aliases, domains)
OpenSIPS Control Panel web interface
PSTN connectivity (dialplan, ACLs, drouting, failover)
Advanced SIP Call Flows (parallel and serial forking, call 
forwarding, call transfer)

SIP presence
SIP Dialog Awareness
Load Balancing with OpenSIPS
OpenSIPS High Availability
SIP NAT traversal
OpenSIPS accounting and billing
SIP security (floods, auth, DNS poisoning, register attacks, TLS 
and SRTP)

OpenSIPS B2BUA ( topology hiding )

More details on:
http://www.opensips.org/Training/EBootcamp

Registration at:
http://ebootcamp.opensips.org/

Questions:
bootc...@opensips.org


Best regards,
Bogdan

--
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OpenSIPS eBootcamp - 2nd of May 2011
OpenSIPS solutions and "know-how"

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Re: [OpenSIPS-Users] parameter not found in module

2011-04-12 Thread Toyima Dias
Thanks maya,

That solved my problem, regards!


2011/4/12 mayamatakeshi 

>
>
>  On Tue, Apr 12, 2011 at 4:57 PM, Toyima Dias  wrote:
>
>> Hello,
>>
>> I'm receiving the following errors on my opensips.log:
>>
>>  *ERROR:core:set_mod_param_regex: parameter  not found in module
>> 
>> *
>> This is what i've added to my opensips.cfg:
>>
>>
>> loadmodule "mi_xmlrpc.so"
>> loadmodule "siptrace.so"
>>
>> # - setting module-specific parameters ---
>>  MODPARAMS FRO SIP TRACE
>> modparam("mi_xmlrpc","port", "8000")
>>
>
>
> Have you tried:
>
> modparam("mi_xmlrpc","port", 8000)
>
>
>
>> modparam("siptrace", "db_url",
>> "mysql://opensips:opensipsrw@localhost/opensips")
>> modparam("siptrace", "trace_flag", 22)
>> modparam("siptrace", "traced_user_avp", "$avp(s:traceuser)")
>>
>> What could be wrong? i'm using *Server:: OpenSIPS (1.6.4-2-notls
>> (i386/linux))*
>>
>
>
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Re: [OpenSIPS-Users] B2B routing

2011-04-12 Thread Anton Zagorskiy
Anca,

one more question :)


Is it possible to make a timer that will represent a lifetime of finished
B2B dialog to catch such SIP requests? (SIP requests for already finished
session).
I just want to say that I'm not interested in such errors (does anyone?) and
I'm afraid of missing 'real' errors.

Also, it will be very helpful, if B2B says that it received request for
already finished session.


WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru



> -Original Message-
> From: users-boun...@lists.opensips.org [mailto:users-
> boun...@lists.opensips.org] On Behalf Of Anca Vamanu
> Sent: Monday, April 11, 2011 6:38 PM
> To: users@lists.opensips.org
> Subject: Re: [OpenSIPS-Users] B2B routing
> 
> Hi Anton,
> 
> 
> On 04/08/2011 03:51 PM, Anton Zagorskiy wrote:
> >
> > It seems that it happens when 2 BYEs received at same time, here is
> > call flow for BYEs (Leg1 is a leg between UA1 and OpenSIPS B2B, Leg2
> > is a leg between OpenSIPS B2B and UA2):
> >
> > 1.
> > Leg 1:  UA1 ->  (BYE)  ->  OpenSIPS ->  (BYE) ->  OpenSIPS B2B; Leg 2:
> > OpenSIPS B2B ->  (BYE) ->  UA2;
> >
> > 2.
> > Leg 2. UA2 ->  (BYE) ->  OpenSIPS B2B ->  (OK) ->  UA2;  Here started
> > OK reply through both legs, which is generated by OpenSIPS - is it
normal?
> >
> At these point the B2BUA considers the dialog closed - because BYE was
> received from both sides, and sends 200OK to both sides. After this, it
> deletes the call record.
> > 3.
> > Leg 1: OpenSIPS B2B ->  (OK) ->  OpenSIPS ->  (OK) ->  UA1;
> >
> > 4.
> > Leg 2: UA2 ->  (OK) ->  OpenSIPS B2B.
> >
> So at this point, the 200 OK will no longer have a match. So that error
> message that you see in the log is not important. Everything functions the
> way it has to.
> > So, I have two questions:
> > 1) OpenSIPS B2B generated OK reply for both legs - is it normal?
> > (dialog module isn't used)
> > 2) The reason of the ERROR message is the OK reply in a stage 4. Is
> > this UA2 bug?
> >
> >
> > Thanks!
> 
> Regards,
> 
> --
> Anca Vamanu
> OpenSIPS Developer
> 
> 
> 
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Re: [OpenSIPS-Users] parameter not found in module

2011-04-12 Thread mayamatakeshi
On Tue, Apr 12, 2011 at 4:57 PM, Toyima Dias  wrote:

> Hello,
>
> I'm receiving the following errors on my opensips.log:
>
>  *ERROR:core:set_mod_param_regex: parameter  not found in module
> 
> *
> This is what i've added to my opensips.cfg:
>
>
> loadmodule "mi_xmlrpc.so"
> loadmodule "siptrace.so"
>
> # - setting module-specific parameters ---
>  MODPARAMS FRO SIP TRACE
> modparam("mi_xmlrpc","port", "8000")
>


Have you tried:
modparam("mi_xmlrpc","port", 8000)



> modparam("siptrace", "db_url",
> "mysql://opensips:opensipsrw@localhost/opensips")
> modparam("siptrace", "trace_flag", 22)
> modparam("siptrace", "traced_user_avp", "$avp(s:traceuser)")
>
> What could be wrong? i'm using *Server:: OpenSIPS (1.6.4-2-notls
> (i386/linux))*
>
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Re: [OpenSIPS-Users] OpenSIPS-CP - Call to undefined method MDB2_Error::setFetchMode()

2011-04-12 Thread Alex Ionescu

Hi Duane,

Have you created the initial user in the opc_admin_privileges  table ?
You have to insert an user - it is written in the INSTALL file.
Anyway, if you did, please check again if the connection details in 
../opensips-cp/config/db.inc.php are the correct ones.
Unfortunatelly, MDB2 is not very "friendly" when reporting errors ... it 
may create confusion.


Let me know if you succeeded!

Regards,
Alex

Anton Zagorskiy wrote:

Hi Duane.

In most cases it means that it can't connect to DB.




From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of
duane.lar...@gmail.com
Sent: Tuesday, April 12, 2011 3:44 AM
To: OpenSIPS users mailling list
Subject: [OpenSIPS-Users] OpenSIPS-CP - Call to undefined method
MDB2_Error::setFetchMode()

I am getting this error with OpenSIPS-CP when I try to log in for the first
time


Fatal error: Call to undefined method MDB2_Error::setFetchMode() in
/var/www/opensips-cp/web/db_connect.php on line 31

I saw that someone else in the mailing list had the exact same error and it
was mentioned that the db.ini.php wasn't configured. As far as I know I
followed the installation instructions. I installed all the pear packages
and edited any files that were supposed to be edited. Not sure what's
causing this. I downloaded with svn.


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[OpenSIPS-Users] parameter not found in module

2011-04-12 Thread Toyima Dias
Hello,

I'm receiving the following errors on my opensips.log:

 *ERROR:core:set_mod_param_regex: parameter  not found in module

*
This is what i've added to my opensips.cfg:


loadmodule "mi_xmlrpc.so"
loadmodule "siptrace.so"

# - setting module-specific parameters ---
 MODPARAMS FRO SIP TRACE
modparam("mi_xmlrpc","port", "8000")
modparam("siptrace", "db_url",
"mysql://opensips:opensipsrw@localhost/opensips")
modparam("siptrace", "trace_flag", 22)
modparam("siptrace", "traced_user_avp", "$avp(s:traceuser)")

What could be wrong? i'm using *Server:: OpenSIPS (1.6.4-2-notls
(i386/linux))*

Regards
*
*
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Re: [OpenSIPS-Users] OpenSIPS-CP - Call to undefined method MDB2_Error::setFetchMode()

2011-04-12 Thread Anton Zagorskiy
Hi Duane.

In most cases it means that it can't connect to DB.




From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of
duane.lar...@gmail.com
Sent: Tuesday, April 12, 2011 3:44 AM
To: OpenSIPS users mailling list
Subject: [OpenSIPS-Users] OpenSIPS-CP - Call to undefined method
MDB2_Error::setFetchMode()

I am getting this error with OpenSIPS-CP when I try to log in for the first
time


Fatal error: Call to undefined method MDB2_Error::setFetchMode() in
/var/www/opensips-cp/web/db_connect.php on line 31

I saw that someone else in the mailing list had the exact same error and it
was mentioned that the db.ini.php wasn't configured. As far as I know I
followed the installation instructions. I installed all the pear packages
and edited any files that were supposed to be edited. Not sure what's
causing this. I downloaded with svn.


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Re: [OpenSIPS-Users] B2B routing

2011-04-12 Thread Anton Zagorskiy
Hi Anca,

Thanks.






WBR, Anton Zagorskiy
VoIP Developer, Oyster Telecom
Phone.: +7 812 601-0666
Fax: +7 812 601-0593
a.zagors...@oyster-telecom.ru
www.oyster-telecom.ru



> -Original Message-
> From: users-boun...@lists.opensips.org [mailto:users-
> boun...@lists.opensips.org] On Behalf Of Anca Vamanu
> Sent: Monday, April 11, 2011 6:38 PM
> To: users@lists.opensips.org
> Subject: Re: [OpenSIPS-Users] B2B routing
> 
> Hi Anton,
> 
> 
> On 04/08/2011 03:51 PM, Anton Zagorskiy wrote:
> >
> > It seems that it happens when 2 BYEs received at same time, here is
> > call flow for BYEs (Leg1 is a leg between UA1 and OpenSIPS B2B, Leg2
> > is a leg between OpenSIPS B2B and UA2):
> >
> > 1.
> > Leg 1:  UA1 ->  (BYE)  ->  OpenSIPS ->  (BYE) ->  OpenSIPS B2B; Leg 2:
> > OpenSIPS B2B ->  (BYE) ->  UA2;
> >
> > 2.
> > Leg 2. UA2 ->  (BYE) ->  OpenSIPS B2B ->  (OK) ->  UA2;  Here started
> > OK reply through both legs, which is generated by OpenSIPS - is it
normal?
> >
> At these point the B2BUA considers the dialog closed - because BYE was
> received from both sides, and sends 200OK to both sides. After this, it
> deletes the call record.
> > 3.
> > Leg 1: OpenSIPS B2B ->  (OK) ->  OpenSIPS ->  (OK) ->  UA1;
> >
> > 4.
> > Leg 2: UA2 ->  (OK) ->  OpenSIPS B2B.
> >
> So at this point, the 200 OK will no longer have a match. So that error
> message that you see in the log is not important. Everything functions the
> way it has to.
> > So, I have two questions:
> > 1) OpenSIPS B2B generated OK reply for both legs - is it normal?
> > (dialog module isn't used)
> > 2) The reason of the ERROR message is the OK reply in a stage 4. Is
> > this UA2 bug?
> >
> >
> > Thanks!
> 
> Regards,
> 
> --
> Anca Vamanu
> OpenSIPS Developer
> 
> 
> 
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Re: [OpenSIPS-Users] uac_replace_from auto mode with to_tag ignores display name

2011-04-12 Thread Razvan Crainea

Hello, Jeff,

Please make sure that the first parameter is not set to "". As you can 
see here[1], if the first parameter of uac_replace_from method is "", it 
removes the Display name.


Regards,
Razvan

[1] http://www.opensips.org/html/docs/modules/devel/uac.html#id292701

On 04/11/2011 07:03 PM, Jeff Pyle wrote:

Hello,

I run uac_replace_from (auto mode) in a branch_route.  Subsequents 
requests for the same dialog (with to_tag, through loose_route) have 
the From URI updated correctly, but the display name isn't.  Is this a 
misconfiguration on my part, by design, or a bug?



- Jeff


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Re: [OpenSIPS-Users] MediaProxy and redirected traffic question

2011-04-12 Thread Saúl Ibarra Corretgé

Hi,

On 5/4/11 9:54 PM, Stylianos Papanastasiou wrote:

Hi everyone,

I am running opensips with mediaproxy and so the media traffic from my
two SIP terminals runs through an intermediate machine. Now, I want to
do some media traffic manipulation so I redirect the incoming traffic to
some port on the intermediate machine using iptables. Assuming my
terminals are 10.0.0.2 and 10.0.0.3 and the media-proxy is 10.0.0.1 -
and the video traffic comes from port 9078 from both terminals I give:

iptables -t nat -A PREROUTING -i eth0  -p udp -s 10.0.0.2 --sport 9078
-j REDIRECT --to-port 4
iptables -t nat -A PREROUTING -i eth0  -p udp -s 10.0.0.3 --sport 9078
-j REDIRECT --to-port 40001



MediaProxy does insert contrack rules to direct the traffic, as you 
know, and the fact that you are altering the routing with iptables may 
have serious implications on what happens from that point on. MediaProxy 
told the kernel to redirect packets coming from 10.0.0.2 and port X to 
10.0.0.3 port Y, but you introduced another routing rule I don't expect 
packets to 'fork'.



--
Saúl Ibarra Corretgé
AG Projects

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