[OpenSIPS-Users] PROBLEM: siptrace not trace incoming CANCEL

[Dragomir Haralambiev]

Hello,

I use latest OpenSips 1.8.2 with siptrace module.

The OpenSips receive CANCEL:

INCOMING CANCEL -> OpenSips --- OUTGOING CANCEL --->

Siptrace module write in DB only OUTGOING CANCEL.

Here part of opensips.cfg

modparam("siptrace", "db_url", "mysql://user:passwd@localhost/opensips")
modparam("siptrace", "trace_on", 1)
modparam("siptrace", "enable_ack_trace", 1)

route{
  sip_trace();
  ..
}

What I do to log Incoming CANCEL ?

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

Re: [OpenSIPS-Users] CANCELd dialogs with top-hiding hanging

[Vlad Paiu]

Hello,

What is the OpenSIPS SVN version and revision that you are using ?
There was recently a bug found, where dialogs would get stuck for 
Cancels that had a To-Tag, and it was fixed in rev 9382 .


Regards,
Vlad

Pe 11/21/2012 9:19 PM, Brett Nemeroff a scris:

Hey All,
I'm doing simple dialog based topo hiding and I noticed that CANCELd 
dialogs are hanging around in state 5. They are starting to stack up.


Here's an example of one of the dialogs.. I kind of thought that when 
the timeout ran down, the dialog would disappear, but it's not:


dialog::  hash=1645:1160860947

state:: 5

user_flags:: 0

timestart:: 0

timeout:: 0

callid:: 69126-1353462109-9996@1.2.3.4 



from_uri:: sip:14387649628@1.2.3.4:5060 



to_uri:: sip:5093111@5.6.7.8 

caller_tag:: 201142122000875167015

caller_contact:: sip:1.2.3.4:5060;transport=udp

callee_cseq:: 0

caller_route_set::

caller_bind_addr:: udp:5.6.7.8:5060 

callee_tag:: 2011411220122012615936129

callee_contact:: sip:9.8.7.6:5060;transport=udp

caller_cseq:: 1


This has been around for about 18 hours! SIP Trace looks pretty normal 
really.



To properly support topo hiding, I've got this at the top of the route 
processing:



if (has_totag()) {

if(is_method("INVITE|ACK|BYE|UPDATE|CANCEL")) {

if(match_dialog()) {

t_relay();

exit;

}

}


I originally didn't include CANCEL in the method check, but it caused 
CANCELs to be ignored by one of the endpoints. Granted, I know the 
endpoints are using switches that arn't known for being very 
compliant. When I added in CANCEL in this check, the sip traces now 
look proper and both sides are happy with the request being 
terminated, but the dialog persists in memory.



Any ideas why these dialogs are not clearing out of opensips?



Thanks,

Brett




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


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

[OpenSIPS-Users] Password in plaintext vs encrypted

[Christian Cambier]

Hello.

When I add a user using the OpenSIPS control panel it stores the password in 
plaintext in the 'password' column

When I add a user using  the /usr/sbin/opensipsctl utility 
   e.g. /usr/sbin/opensipsctl add 1009 99
it stores the password encrypted in the 'ha1'  column

never mind whether   
store_plaintext_pw is set to 0 or 1 
in /etc/opensips/opensipsctlrc

How can I have passwords stored in hashed form with the Control Panel as well?

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

[OpenSIPS-Users] CANCELd dialogs with top-hiding hanging

[Brett Nemeroff]

Hey All,
I'm doing simple dialog based topo hiding and I noticed that CANCELd
dialogs are hanging around in state 5. They are starting to stack up.

Here's an example of one of the dialogs.. I kind of thought that when the
timeout ran down, the dialog would disappear, but it's not:

dialog::  hash=1645:1160860947

state:: 5

user_flags:: 0

timestart:: 0

timeout:: 0

callid:: 69126-1353462109-9996@1.2.3.4

from_uri:: sip:14387649628@1.2.3.4:5060

to_uri:: sip:5093111@5.6.7.8

caller_tag:: 201142122000875167015

caller_contact:: sip:1.2.3.4:5060;transport=udp

callee_cseq:: 0

caller_route_set::

caller_bind_addr:: udp:5.6.7.8:5060

callee_tag:: 2011411220122012615936129

callee_contact:: sip:9.8.7.6:5060;transport=udp

caller_cseq:: 1


This has been around for about 18 hours! SIP Trace looks pretty normal
really.


To properly support topo hiding, I've got this at the top of the route
processing:


if (has_totag()) {

if(is_method("INVITE|ACK|BYE|UPDATE|CANCEL")) {

if(match_dialog()) {

t_relay();

exit;

}

}


I originally didn't include CANCEL in the method check, but it caused
CANCELs to be ignored by one of the endpoints. Granted, I know the
endpoints are using switches that arn't known for being very compliant.
When I added in CANCEL in this check, the sip traces now look proper and
both sides are happy with the request being terminated, but the dialog
persists in memory.


Any ideas why these dialogs are not clearing out of opensips?



Thanks,

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

Re: [OpenSIPS-Users] how to Register with OpenSIPS?

[Muhammad Shahzad]

Well, generally you use opensip for registration and call routing and PABX
for actual media services. Have a look at below tutorial,

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

Thank you.


On Thu, Nov 15, 2012 at 1:54 PM, Christian Cambier <
christian.camb...@gmail.com> wrote:

> Hi.
>
> Here's what i try to do: I have a SipPhone with extension 5006
> Without proxy i register at a PBX with address (10.1.2.3). Ok, no problem
>
> What I'm struggling with is, how can i have this phone register at that
> PBX when every SIP request has to pass via a OpenSIPS proxy first.
> Somehow the proxy must know how to forward REGISTER messages to the PBX
> no? I'd say, It must know the existence of the PBX.
> Or don't I need the PBX for REGISTER?
> But surely, INVITE must pass the PBX no?
>
> Anyway, how do I have a cooporate OpenSIPS proxy with a PBX?
> Where to configure what?
>
> Please help!!!
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] OpenSIPS Control Panel limitation with table dialplan for more than 40k rows.

[Muhammad Shahzad]

Its most like your web server issue, you need to increase output buffer
size. If using Apache check error_log file, it will tell you exactly what
is happening.

Thank you.


2012/11/7 Miguel J. 

> **
> Dear Opensips lists:
>
> I've found a trouble with the OpenSips Control Panel tool when the
> dialplan table has more than 4 rows.
>
> If this table contains up to 40,000 rows, the content is provided in
> the web tool and everything works fine. If this table contains a record
> more, the tool leaves the panel empty, as I show in the attached document.
>
> Can I configure or adapt the tool to avoid this limitation?
>
> Thank you very much and best regards
>   --
>
>
>
> -
> Sus datos de carácter personal (nombre, apellidos, dirección postal y de
> correo electrónico, etc.) son tratados para la gestión de su relación con
> la Entidad, así como para el envío de información sobre nuestra actividad y
> la de terceros relacionadas con la actividad de Consulting Smartic
> Solutions, S.L., CIF: B85130037, C/Pº de la Castellana, 135, 7ª planta,
> 28046 Madrid. Usted puede ejercer sus derechos de acceso, rectificación,
> cancelación y oposición dirigiéndose por escrito, con copia de un documento
> que acredite su identidad, a la dirección info (arroba) smartic.es.
> Este mensaje puede contener información confidencial. Si usted no es su
> destinatario, no debe leerlo, copiarlo, distribuirlo, ni hacer uso de la
> información que contiene. En este caso, por favor, llámenos o
> comuníquenoslo por escrito y borre este mensaje de su sistema.
> -
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
<>___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Opensips don't send Cancel.

[Muhammad Shahzad]

Well, custom headers module parameter is the right place for your needs, if
its not working then its most likely a bug and you should report it.

Thank you.


On Wed, Nov 21, 2012 at 6:02 PM, Jorge Henrique Pinho <
jorge-h-pi...@ext.ptinovacao.pt> wrote:

> Hi, thank for your feedback.
> Custom headers was my first approach, but the problem remains, the
> User-Agent header pass from the dialog of one side to the other side but
> the value is changed anyway.
>
> Is there a way that i can define specific headers to be untouched by b2b
> module or anywhere in the routines where i already see the modified message
> and have permissions to manipulate it?
>
> Kind regards,
>
> Jorge Pinho
>
> 
> From: users-boun...@lists.opensips.org [users-boun...@lists.opensips.org]
> On Behalf Of Muhammad Shahzad [shaherya...@gmail.com]
> Sent: 21 November 2012 14:53
> To: OpenSIPS users mailling list
> Subject: Re: [OpenSIPS-Users] Opensips don't send Cancel.
>
> No, don't temper with opensips generated request in local route, it will
> result in unexpected behaviour as you have already observed, right right
> way to do is define custom headers that should be passed from one leg to
> other leg, as described here,
>
> http://www.opensips.org/html/docs/modules/1.8.x/b2b_logic.html#id250020
>
> The purpose of local route is to do things that do not modify generated
> request in general (specially in case of B2BUA never modify the request).
> You can use this route e.g. to trigger a billing event, or cdr or some db
> operation etc. etc.
>
> Thank you.
>
>
> On Wed, Nov 21, 2012 at 12:51 PM, Jorge Henrique Pinho <
> jorge-h-pi...@ext.ptinovacao.pt>
> wrote:
> Good morning.
> I have installed opensips-1.7.2-2.el5 and I am using b2b module and
> encountering a strange behavior.
> B2b module changes the original User Agent header and I need this
> information to be preserved; so in the route script I used an avp to store
> the original User Agent value, call b2b_init_request and on local_route
> script I use remove_hf and append_hf functions to replace the User Agent
> header with the original one.
>
> The behavior is : If I want to cancel the call, opensips receives a Cancel
> message, replies with a 200 canceling but never forwards the Cancel message.
> The behavior is not deterministic, sometimes opensips forwards the Cancel,
> and sometimes it does not.
>
> When I don't use the remove_hf and append_hf functions, opensips behaves
> as expected, and always forwards the Cancel.
>
> Kind regards
>
> Jorge Pinho
>
>
> Jorge Pinho
> Analyst/Developer
> Network Platforms and Multimedia Solutions
> Multimedia Division
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Opensips don't send Cancel.

[Jorge Henrique Pinho]

Hi, thank for your feedback.
Custom headers was my first approach, but the problem remains, the User-Agent 
header pass from the dialog of one side to the other side but the value is 
changed anyway.

Is there a way that i can define specific headers to be untouched by b2b module 
or anywhere in the routines where i already see the modified message and have 
permissions to manipulate it?

Kind regards,

Jorge Pinho


From: users-boun...@lists.opensips.org [users-boun...@lists.opensips.org] On 
Behalf Of Muhammad Shahzad [shaherya...@gmail.com]
Sent: 21 November 2012 14:53
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Opensips don't send Cancel.

No, don't temper with opensips generated request in local route, it will result 
in unexpected behaviour as you have already observed, right right way to do is 
define custom headers that should be passed from one leg to other leg, as 
described here,

http://www.opensips.org/html/docs/modules/1.8.x/b2b_logic.html#id250020

The purpose of local route is to do things that do not modify generated request 
in general (specially in case of B2BUA never modify the request). You can use 
this route e.g. to trigger a billing event, or cdr or some db operation etc. 
etc.

Thank you.


On Wed, Nov 21, 2012 at 12:51 PM, Jorge Henrique Pinho 
mailto:jorge-h-pi...@ext.ptinovacao.pt>> wrote:
Good morning.
I have installed opensips-1.7.2-2.el5 and I am using b2b module and 
encountering a strange behavior.
B2b module changes the original User Agent header and I need this information 
to be preserved; so in the route script I used an avp to store the original 
User Agent value, call b2b_init_request and on local_route script I use 
remove_hf and append_hf functions to replace the User Agent header with the 
original one.

The behavior is : If I want to cancel the call, opensips receives a Cancel 
message, replies with a 200 canceling but never forwards the Cancel message.
The behavior is not deterministic, sometimes opensips forwards the Cancel, and 
sometimes it does not.

When I don't use the remove_hf and append_hf functions, opensips behaves as 
expected, and always forwards the Cancel.

Kind regards

Jorge Pinho


Jorge Pinho
Analyst/Developer
Network Platforms and Multimedia Solutions
Multimedia Division
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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

Re: [OpenSIPS-Users] Errors in OpenSIPS

[Alex Ionescu]

Hi,

Most probably not. OpenSIPS usually does updates/inserts on this table. 
Only if OpenSIPS is configured to restore it's dialogs from DB in case 
of restart then you might want to make a test just to be sure it won't 
complain.


Regards,
Alex Ionescu
On 11/15/2012 01:20 PM, Jorge Ortea wrote:

Hello,

can someone said me if really modify dialog table and allow 
callee_contract field to be NULL on DB could break anything??


Thanks.
Regards.


Date: Tue, 13 Nov 2012 14:48:51 +0100
From: shaherya...@googlemail.com
To: users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Errors in OpenSIPS

Its db schema problem. The field dialog.callee_contact is set as "NOT 
NULL", while you are getting NULL contact from upstream (most likely 
in BYE from callee side, as some old PSTN gateways running at many 
termination providers do not add "Contact" header in BYE request). 
Please check SIP trace and confirm or attach that here for our review.


For a workaround you can modify dialog table and allow callee_contract 
field to be NULL. However i am not sure if this won't break anything else.


Thank you.


On Mon, Nov 12, 2012 at 1:52 PM, Jorge Ortea > wrote:


Hello,

I keep getting these errors.

My opensips version is 1.6.4-2-tls and my mysql version is 5.1.34

Would it be better to change the version of opensips?

What can mean  "Column 'callee_contact' cannot be null" ??

Thanks.
Regards.


From: dar...@hotmail.com 
To: users@lists.opensips.org 
Date: Fri, 9 Nov 2012 08:30:03 +0100

Subject: Re: [OpenSIPS-Users] Errors in OpenSIPS

Hello Vlad,

You are right, I had searched for ERROR, but before each Error
message I got this:

Nov  5 10:24:19 hgt-tero45
/usr/local/opensips/sbin/opensips[12562]:
CRITICAL:db_mysql:wrapper_single_mysql_stmt_execute: driver error
(1048): Column 'callee_contact' cannot be null
Nov  5 10:24:19 hgt-tero45
/usr/local/opensips/sbin/opensips[12562]:
ERROR:dialog:update_dialog_dbinfo: could not add another dialog to db

Why could this be happening?

Thanks.
Regards.





Date: Thu, 8 Nov 2012 19:21:45 +0200
From: vladp...@opensips.org 
To: users@lists.opensips.org 
Subject: Re: [OpenSIPS-Users] Errors in OpenSIPS

Hello Jorge,

Do you see any other errors from the DB level in the OpenSIPS logs
? What DB backend are you using ?
It seems that for some reason, the dialog DB inserts are failing,
but more ERROR messages should appear in such a case from the
db_mysql module..

Regards,

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com  



On 11/08/2012 07:10 PM, Ali Pey wrote:

I think it's telling you it can't add another dialog to db. I
am almost certain that's what's happening.

Regards,
Ali Pey



On Thu, Nov 8, 2012 at 4:22 AM, Jorge Ortea
mailto:dar...@hotmail.com>> wrote:

Hi all,

I am getting the following errors:


Nov  7 11:47:19 hgt-tero45
/usr/local/opensips/sbin/opensips[12582]:
ERROR:dialog:update_dialog_dbinfo: could not add another
dialog to db
Nov  7 11:47:19 hgt-tero45
/usr/local/opensips/sbin/opensips[12548]:
ERROR:dialog:update_dialog_dbinfo: could not add another
dialog to db
Nov  7 11:47:20 hgt-tero45
/usr/local/opensips/sbin/opensips[12558]:
ERROR:dialog:update_dialog_dbinfo: could not add another
dialog to db
Nov  7 11:57:32 hgt-tero45
/usr/local/opensips/sbin/opensips[12554]:
ERROR:dialog:update_dialog_dbinfo: could not add another
dialog to db
Nov  7 11:57:32 hgt-tero45
/usr/local/opensips/sbin/opensips[12578]:
ERROR:dialog:update_dialog_dbinfo: could not add another
dialog to db
Nov  7 11:59:07 hgt-tero45
/usr/local/opensips/sbin/opensips[12556]:
ERROR:dialog:update_dialog_dbinfo: could not add another
dialog to db
Nov  7 11:59:07 hgt-tero45
/usr/local/opensips/sbin/opensips[12574]:
ERROR:dialog:update_dialog_dbinfo: could not add another
dialog to db
Nov  7 12:23:23 hgt-tero45
/usr/local/opensips/sbin/opensips[12580]:
ERROR:dialog:update_dialog_dbinfo: could not add another
dialog to db
Nov  7 12:23:24 hgt-tero45
/usr/local/opensips/sbin/opensips[12544]:
 

Re: [OpenSIPS-Users] Load Balancer Issue

[Bogdan-Andrei Iancu]

Nilanjan,

The caller script is broken - in ACK, is should be ROUTE hdrs where you 
have the RECORD-ROUTE ones :) .ACK should look like:


U 2012/11/01 11:19:02.006514 X.X.X.23:5080 -> X.X.X.206:5060
ACK sip:X.X.X.5:5070;transport=UDP SIP/2.0.
Route: 
,.

Via: SIP/2.0/UDP X.X.X.23:5080;branch=z9hG4bK-31168-1-5.
From: sipp ;tag=31168SIPpTag001.
To: sut ;tag=30500SIPpTag011.
Call-ID: 1-31168@X.X.X.23.
CSeq: 1 ACK.
Contact: sip:sipp@X.X.X.23:5080.
Max-Forwards: 70.
Subject: Performance Test.
Content-Length: 0.

I guess it is an err in your sipp scenario.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 11/09/2012 07:24 PM, Nilanjan Banerjee wrote:

Hi Bogdan,

  Thanks for your reply. The answer to both your questions is yes. 
Sample of OK and ACK at the caller as follows:


#
U 2012/11/01 11:19:02.006375 X.X.X.206:5060 -> X.X.X.23:5080
SIP/2.0 200 OK.
Record-Route: , 
.

Via: SIP/2.0/UDP X.X.X.23:5080;branch=z9hG4bK-31168-1-0.
From: sipp ;tag=31168SIPpTag001.
To: sut ;tag=30500SIPpTag011.
Call-ID: 1-31168@X.X.X.23.
CSeq: 1 INVITE.
Contact: .
Content-Type: application/sdp.
Content-Length:   137.
.
v=0.
o=user1 53655765 2353687637 IN IP4 X.X.X.5.
s=-.
c=IN IP4 X.X.X.5.
t=0 0.
m=audio 6000 RTP/AVP 0.
a=rtpmap:0 PCMU/8000.

#
U 2012/11/01 11:19:02.006514 X.X.X.23:5080 -> X.X.X.206:5060
ACK sip:X.X.X.5:5070;transport=UDP SIP/2.0.
Record-Route: 
,.

Via: SIP/2.0/UDP X.X.X.23:5080;branch=z9hG4bK-31168-1-5.
From: sipp ;tag=31168SIPpTag001.
To: sut ;tag=30500SIPpTag011.
Call-ID: 1-31168@X.X.X.23.
CSeq: 1 ACK.
Contact: sip:sipp@X.X.X.23:5080.
Max-Forwards: 70.
Subject: Performance Test.
Content-Length: 0.
.

Thanks,
Nilanjan.

On Fri, Nov 9, 2012 at 10:29 PM, Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:


Hi Nilanjan,

Check in the trace if :
1) the 200 OK getting back to the caller has 2 RR headers (one
from Proxy and one from LB).

2) the ACK from caller (before LB) has 2 Route headers, one
pointing to LB, next to Proxy.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 11/09/2012 05:01 PM, Nilanjan Banerjee wrote:

Hi Bogdan,

  Thanks a lot for your suggestion and sorry for the delay in
getting back with this...I tried the following configuration as
you have suggested for the Load Balancer and the Proxy:

__
Load Balancer:
__

route{
if (!mf_process_maxfwd_header("3")) {
sl_send_reply("483","looping");
exit;
}

if (!has_totag()) {
# initial request
record_route();
} else {
# sequential request -> obey Route indication
loose_route();
t_relay();
exit;
}

# detect resources and do load balancing

 load_balance("1","sc");

# LB function returns negative if no suitable destination
(for requested resources) is found,
# or if all destinations are full
if ($retcode<0) {
 sl_send_reply("500","Service full");
 exit;
}

xlog("Selected destination is: $du\n");

# send it out
if (!t_relay()) {
sl_reply_error();
}
}

__
Proxy
__

route{

if (!has_totag()) {
# initial request
record_route();
} else {
# sequential request -> obey Route indication
loose_route();
}

if (!t_relay()) {
 #   xlog("L_ERR","sl_reply_error\n");
sl_reply_error();
}

}

However, I am still getting the same error - basically the ACK
and the BYE messages are skipping the Proxy and the response to
the BYE is sent to the Proxy. Here are the sample ACK and BYE for
the following setup I am using:

X.X.X.23:5080 --> X.X.X.206:5060 --> X.X.X.8:5060 --> X.X.X.5:5070
(sipp UAC)   --> (Load Balancer) -->  (Proxy) -->
(sipp UAS)

#
U 2012/11/01 11:19:22.901990 X.X.X.206:5060 -> X.X.X.5:5070
ACK sip:X.X.X.5:5070;transport=UDP SIP/2.0.
Record-Route:
,.
Via: SIP/2.0/UDP X.X.X.206;branch=z9hG4bK0112.20fe162.2.
Via: SIP/2.0/UDP X.X.X.23:5080;branch=z9hG4bK-31168-5-5.
From: sipp ;tag=31168SIPpTag005.
To: sut ;tag=30500SIPpTag015.
Call-ID: 5-31168@X.X.X.23 .
CSeq: 1 ACK.
Contact: sip:sipp@X.X.X.23:5080.
Max-Forwards: 69.
Subject: Performance Test.
Content-Length: 0.
.

#
U 2012/11/01 11:19:22.934118 X.X.X.23:5080 -> X.X.X.206:5060
BYE sip:X.X.X.5:5070;transport=UDP SIP/2.0.
Record-Route:
,.
Via: SIP/2.0/UDP X.X.X.23:5080

[OpenSIPS-Users] how to Register with OpenSIPS?

[Christian Cambier]

Hi.

Here's what i try to do: I have a SipPhone with extension 5006
Without proxy i register at a PBX with address (10.1.2.3). Ok, no problem

What I'm struggling with is, how can i have this phone register at that PBX
when every SIP request has to pass via a OpenSIPS proxy first.
Somehow the proxy must know how to forward REGISTER messages to the PBX
no? I'd say, It must know the existence of the PBX.
Or don't I need the PBX for REGISTER?
But surely, INVITE must pass the PBX no?

Anyway, how do I have a cooporate OpenSIPS proxy with a PBX?
Where to configure what?

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

Re: [OpenSIPS-Users] Opensips don't send Cancel.

[Muhammad Shahzad]

No, don't temper with opensips generated request in local route, it will
result in unexpected behaviour as you have already observed, right right
way to do is define custom headers that should be passed from one leg to
other leg, as described here,

http://www.opensips.org/html/docs/modules/1.8.x/b2b_logic.html#id250020

The purpose of local route is to do things that do not modify generated
request in general (specially in case of B2BUA never modify the request).
You can use this route e.g. to trigger a billing event, or cdr or some db
operation etc. etc.

Thank you.


On Wed, Nov 21, 2012 at 12:51 PM, Jorge Henrique Pinho <
jorge-h-pi...@ext.ptinovacao.pt> wrote:

> Good morning.
> I have installed opensips-1.7.2-2.el5 and I am using b2b module and
> encountering a strange behavior.
> B2b module changes the original User Agent header and I need this
> information to be preserved; so in the route script I used an avp to store
> the original User Agent value, call b2b_init_request and on local_route
> script I use remove_hf and append_hf functions to replace the User Agent
> header with the original one.
>
> The behavior is : If I want to cancel the call, opensips receives a Cancel
> message, replies with a 200 canceling but never forwards the Cancel message.
> The behavior is not deterministic, sometimes opensips forwards the Cancel,
> and sometimes it does not.
>
> When I don't use the remove_hf and append_hf functions, opensips behaves
> as expected, and always forwards the Cancel.
>
> Kind regards
>
> Jorge Pinho
>
>
> Jorge Pinho
> Analyst/Developer
> Network Platforms and Multimedia Solutions
> Multimedia Division
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] Opensips don't send Cancel.

[Jorge Henrique Pinho]

Good morning.
I have installed opensips-1.7.2-2.el5 and I am using b2b module and 
encountering a strange behavior.
B2b module changes the original User Agent header and I need this information 
to be preserved; so in the route script I used an avp to store the original 
User Agent value, call b2b_init_request and on local_route script I use 
remove_hf and append_hf functions to replace the User Agent header with the 
original one.

The behavior is : If I want to cancel the call, opensips receives a Cancel 
message, replies with a 200 canceling but never forwards the Cancel message.
The behavior is not deterministic, sometimes opensips forwards the Cancel, and 
sometimes it does not.

When I don't use the remove_hf and append_hf functions, opensips behaves as 
expected, and always forwards the Cancel.

Kind regards

Jorge Pinho


Jorge Pinho
Analyst/Developer
Network Platforms and Multimedia Solutions 
Multimedia Division 
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] problem with domain

[Christian Cambier]

Hi.

I have a UAC that can ping the proxy on 10.0.4.34

On the proxy I've a user 100 in subscriber table
100  | 10.0.4.34 |  | 1...@cca.com  

registering the UAC with settings:
extension 100 
proxy 10.0.4.34
works fine

Now, on the client I can as well ping the proxy with sip.cca.com
which displays 10.0.4.34

If I change the domain for user 100 on the proxy
100  | sip.cca.com|  | 1...@cca.com  

then I get a SIP error when registering
'SIP/2.0 403 Preload Route denied 
Via: SIP/2.0/TCP
10.0.46.1:54558;received=10.0.46.1;rport=54558;branch=z9hG4bKPj692483511
1304cc3af945bb461b9a42e 
From: "cid100"
;tag=46d7102eb38241c08d49913374fe6055 
To: "cid100"
;tag=c97b4d1cb1f3d0da549e06a8d482ef63.d0da 
Call-ID: d672158176c54358a6d9d1ceccea1222 
CSeq: 48418 REGISTER 
Server: OpenSIPS (1.8.0-notls (x86_64/linux)) 
Content-Length: 0

What am I missing?

thx
Chris

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

Re: [OpenSIPS-Users] Question about Parallel Forking

[SamyGo]

Hi,
I think Its done now. This is easy.
if your call is already going out to first user and then you want it to go
in parallel to second destination too then all you need it to do just
before hitting the t_relay() is

$branch= "sip:newdestincation@secondip:port";

and you'll see it go to both destinations in parallel.

Regards,
Gohar

On Mon, Nov 19, 2012 at 7:45 PM, spady  wrote:

> E1   SIP   Fork   |>
> PBX1
> PSTN->Mediant 1000--->OpenSIPS|
>
> |>
> PBX2
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

[OpenSIPS-Users] SIP/2.0 401 Unauthorized

[Christian Cambier]

Hello.

My password is stored in plaintext in the password column
++--+---+--+---+-+--+---
---+

| id | username | domain| password | email_address | ha1 | ha1b |
rpid |

++--+---+--+---+-+--+---
---+

|  6 | 1006 | 10.0.4.34 | 5| 1...@sip.com  | |  |
NULL |

++--+---+--+---+-+--+---
---+

 I assume that the  setting I've changed in /etc/opensips/opensipsctlrc
from 
  store_plaintext_pw=1
to 
  store_plaintext_pw=0
hasn't been taken into consideration.

How do I activate this setting, what service that reads this config file
do I have to restart so it'll use this changed setting?

thx
Chris

 

From: Christian Cambier 
Sent: dinsdag 20 november 2012 18:01
To: 'OpenSIPS users mailling list'
Subject: SIP/2.0 401 Unauthorized

 

Hello.

Upon registering a UAC I keep on getting a "SIP/2.0 401 Unauthorized" 
The SIP challenge-response is being executed but I assume there's still
an issue with plain/hashed passwords

In the cfg I specify
   modparam("auth_db", "calculate_ha1", 0)

   modparam("auth_db", "password_column", "ha1")


In /etc/opensips/opensipsctlrc I have changed setting 
  store_plaintext_pw=1
to 
  store_plaintext_pw=0

Apart from 
   service opensips restart
do I have to restart anything else or specify another setting somewhere?

thx
Chris

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