Re: [OpenSIPS-Users] How to make OpenSIPS send SIP BYE, by configuration, , before dialog timout?

2016-11-11 Thread Răzvan Crainea 

Hi, Rodrigo!

Sorry, I've just seen the message, I've missed it earlier.

As far as I understand, OpenSIPS is listening on two interfaces: 
127.0.01:5060 and 192.168.0.101:5060. Is the UPDATE coming on the same 
TCP connection as the initial one? Or the client opens a new connection 
for it, over the PUBLIC interface? Could you send over (privately) a 
PCAP trace?
Also, you'd probably need to make sure that you call fix_nated_contact() 
and force_rport() on the UPDATE request. Also, are you setting the 
tcp_accept_aliases[1] or force_tcp_alias()[2] in your script?


[1] http://www.opensips.org/Documentation/Script-CoreParameters-2-2#toc95
[2] 
http://www.opensips.org/Documentation/Script-CoreFunctions-2-2#force_tcp_alias


Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 11/10/2016 09:59 PM, Rodrigo Pimenta Carvalho wrote:


Hi Razvan.


I answered your questions yesterday.


I'm not sure if you saw my message.


Best regards.




RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



*De:* users-boun...@lists.opensips.org 
 em nome de Răzvan Crainea 


*Enviado:* quarta-feira, 9 de novembro de 2016 08:29
*Para:* users@lists.opensips.org
*Assunto:* Re: [OpenSIPS-Users] How to make OpenSIPS send SIP BYE, by 
configuration, , before dialog timout?

Hi, Rodrigo!

The only HACK that I can think of is when you get the BYE message, set 
the dialog timeout to 0, match it against the dialog, and then drop 
the message. OpenSIPS will behave as if the dialog expired in that moment.


However, you seem to have a flow logic - most likely the Contact 
header in the BYE is not correct. Could you send us a trace to help 
you figure out what the problem is? Also, did you try to validate the 
message against the dialog[1] and fix it accordingly[2]?


[1] http://www.opensips.org/html/docs/modules/2.2.x/dialog#id295894
[2] http://www.opensips.org/html/docs/modules/2.2.x/dialog#id295982

Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
Home — OpenSIPS Solutions 
www.opensips-solutions.com
OpenSIPS is a mature Open Source implementation of a SIP server. 
OpenSIPS is more than a SIP proxy/router as it includes 
application-level functionalities.


On 11/08/2016 10:07 PM, Rodrigo Pimenta Carvalho wrote:


Hi.


Dialogs in my OpenSIPS is programmed to finish after 60 seconds. 
(timeout = 1 minute).


So, when 2 peers are in a dialog, OpenSIPS sends SIP BYE to both 
peers, automatically after 60 seconds.



Is it possible to make OpenSIPS send this exact kind of SIP BYE to 
both peers, before the dialog timeout? I mean, in a configured way 
(opensips.cfg)?



When OpenSISP sends SIP BYE automatically, both peers receive the SIP 
BYE correctly.


However, when a peer sends SIP BYE, it reaches the OpenSIPS, but 
OpenSIPS is unable to forward this SIP BYE. Due to some unknown 
reason, in this moment there is no open socket to communicate with 
such peer. That is why I would like to make OpenSIPS send 'its own' 
SIP BYE, and see if such idea will simulate a normal situation, until 
I discover why there is a socket problem.



Any hint will be very helpful!


Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979


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Re: [OpenSIPS-Users] usage of setdsturi

2016-11-11 Thread Răzvan Crainea 

Hi, Agalya!

The setdsturi() function only accepts strings as parameters, not 
pseudo-variables[1]. As Ben suggested, the $du pseudo-variable is more 
flexible and recommended.


[1] http://www.opensips.org/Documentation/Script-CoreFunctions-2-2#toc49

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 11/10/2016 11:58 PM, Newlin, Ben wrote:


I would recommend just using $du. [1]

$du = “sip:” + $var(Fqdn) + “:5060”;

[1] http://www.opensips.org/Documentation/Script-CoreVar-2-2#toc35

Ben Newlin

*From: * on behalf of "Ramachandran, 
Agalya (Contractor)" 

*Reply-To: *OpenSIPS users mailling list 
*Date: *Thursday, November 10, 2016 at 4:35 PM
*To: *OpenSIPS users mailling list 
*Subject: *[OpenSIPS-Users] usage of setdsturi

Hi team,

I have a question in usage of setdsturi().

When I hardcode the uri in the function, such as 
setdsturi(“sip:t...@test.com:5060”) – this works.


But why I try to use script variable, it complains as bad_uri.

$var(test) = "sip:"+$var(Fqdn)+ ":5060";

setdsturi("$var(test)");

How do I setdsturi() dynamically, with the value in script variable 
and not by hardcoding?


Regards,
Agalya



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Re: [OpenSIPS-Users] How to make OpenSIPS send SIP BYE, by configuration, , before dialog timout?

2016-11-11 Thread Rodrigo Pimenta Carvalho 
Hi Rasvan Crainea.


Thank you very much for the reply!

You gave me new points to  be checked and understood.

I will work for a while in such points and then I will give you a feedback.

So, wait for my next post, please.


Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



De: users-boun...@lists.opensips.org  em nome 
de Răzvan Crainea 
Enviado: sexta-feira, 11 de novembro de 2016 11:30
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] How to make OpenSIPS send SIP BYE, by 
configuration, , before dialog timout?

Hi, Rodrigo!

Sorry, I've just seen the message, I've missed it earlier.

As far as I understand, OpenSIPS is listening on two interfaces: 127.0.01:5060 
and 192.168.0.101:5060. Is the UPDATE coming on the same TCP connection as the 
initial one? Or the client opens a new connection for it, over the PUBLIC 
interface? Could you send over (privately) a PCAP trace?
Also, you'd probably need to make sure that you call fix_nated_contact() and 
force_rport() on the UPDATE request. Also, are you setting the 
tcp_accept_aliases[1] or force_tcp_alias()[2] in your script?

[1] http://www.opensips.org/Documentation/Script-CoreParameters-2-2#toc95
[2] 
http://www.opensips.org/Documentation/Script-CoreFunctions-2-2#force_tcp_alias

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

Home — OpenSIPS Solutions
www.opensips-solutions.com
OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is 
more than a SIP proxy/router as it includes application-level functionalities.

On 11/10/2016 09:59 PM, Rodrigo Pimenta Carvalho wrote:

Hi Razvan.


I answered your questions yesterday.


I'm not sure if you saw my message.


Best regards.




RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



De: users-boun...@lists.opensips.org 
 em 
nome de Răzvan Crainea 
Enviado: quarta-feira, 9 de novembro de 2016 08:29
Para: users@lists.opensips.org
Assunto: Re: [OpenSIPS-Users] How to make OpenSIPS send SIP BYE, by 
configuration, , before dialog timout?

Hi, Rodrigo!

The only HACK that I can think of is when you get the BYE message, set the 
dialog timeout to 0, match it against the dialog, and then drop the message. 
OpenSIPS will behave as if the dialog expired in that moment.

However, you seem to have a flow logic - most likely the Contact header in the 
BYE is not correct. Could you send  us a trace to help you figure out what the 
problem is? Also, did you try to validate the message against the dialog[1] and 
fix it accordingly[2]?

[1] http://www.opensips.org/html/docs/modules/2.2.x/dialog#id295894
[2] http://www.opensips.org/html/docs/modules/2.2.x/dialog#id295982

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

Home — OpenSIPS Solutions
www.opensips-solutions.com
OpenSIPS is a mature Open Source implementation of a SIP server. OpenSIPS is 
more than a SIP proxy/router as it includes application-level functionalities.

On 11/08/2016 10:07 PM, Rodrigo Pimenta Carvalho wrote:

Hi.


Dialogs in my OpenSIPS is programmed to finish after 60 seconds. (timeout = 1 
minute).

So, when 2 peers are in a dialog, OpenSIPS sends SIP BYE to both peers, 
automatically after 60 seconds.


Is it possible to make OpenSIPS send this exact kind of SIP BYE to both peers, 
before the dialog timeout? I mean, in a configured way (opensips.cfg)?


When OpenSISP sends SIP BYE automatically, both peers receive the SIP BYE 
correctly.

However, when a peer sends SIP BYE, it reaches the OpenSIPS, but OpenSIPS is 
unable to forward this SIP BYE. Due to some unknown reason, in this moment 
there is no open socket to communicate with such peer. That is why I would like 
to make OpenSIPS send 'its own' SIP BYE, and see if such idea will simulate a 
normal situation, until I discover why there is a socket problem.


Any hint will be very helpful!


Best regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979



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[OpenSIPS-Users] Topology_Hiding adding extra VIA header

2016-11-11 Thread SamyGo 
Hi,

I'm using OpenSIPS 2.2.1 version and I'm facing a weird situation where
OpenSIPS is adding a duplicated VIA header to the 200 OK, This only happens
when I've topology_hiding() engaged into the call.

The scenario is very simple; two users making call to each other on the
same OpenSIPS but with topology_hiding(). As a consequence of this double
VIA the caller device doesn't trigger the ACK and hence we don't get media
stream established between devices.


*WITH TOPOLOGYHIDING:*
SIP/2.0 200 OK
Via: SIP/2.0/TLS 10.1.10.51:59231;received=7X.XX.XX.X7;rport=59231;branch=
z9hG4bK-607165482-63
Via: SIP/2.0/TLS 10.1.10.51:59231;received=7X.XX.XX.X7;rport=59231;branch=
z9hG4bK-607165482-63
CSeq: 1 INVITE
...


*WITHOUT TOPOHIDING:*
SIP/2.0 200 OK
Via: SIP/2.0/TLS 10.1.10.51:59223;received=7X.XX.XX.X7;rport=59223;branch=
z9hG4bK-607166212-58
CSeq: 1 INVITE


The only difference between the two scenarios is the function
topology_hiding(); is commented out.

It seems like a bug to me, can anyone guide me here validate this.

*OpenSIPS Version:*
version: opensips 2.2.1 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC,
FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
git revision: 68ace2e
main.c compiled on 18:34:37 Sep 28 2016 with gcc 4.8


Thanks,
Sammy
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