Re: [OpenSIPS-Users] Mais um Iniciante..

2011-09-07 Thread Daniel Cardoso Alves
Hello Jean,

Send email for this list only in english.

Documentation in portuguese about Opensips isnt easy to find.

I suggest you start reading http://www.opensips.org/Resources/Documentation 
(Only in english)


Best Regards,

===


Boa Noite Jean,

Esta lista é apenas em inglês. 

Material em português será dificil. Sugiro que você comece olhando a 
documentação do sistema: http://www.opensips.org/Resources/Documentation 
(Somente em inglês)




Atenciosamente,


Daniel

 
 
 
 
 
 
 
 
 
 
 
 
 



De: Jean Carlos Coelho 
Para: users@lists.opensips.org
Enviadas: Quarta-feira, 7 de Setembro de 2011 21:18
Assunto: [OpenSIPS-Users] Mais um Iniciante..


Boa noite senhores(as), gostaria de saber se existe um bom material sobre o 
serviço em português, algo como instalação e configurações simples, pelo menos 
até que eu consiga uma boa base para experimentar coisas mais avançadas..

O que eu preciso:

Instalação prática (Debian) - consegui baixando o pacote SVN e gerando os .debs 
porém não sei se devo confiar nestas configurações
Conexões com MySQL ou PostgreSQL
Integração entre dois servidores Asterisk e o OpenSIPS (roteamento de 
ramais/voz)

Qualquer ajuda é bem vinda!

Abraços! :)
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[OpenSIPS-Users] Mais um Iniciante..

2011-09-07 Thread Jean Carlos Coelho
Boa noite senhores(as), gostaria de saber se existe um bom material sobre o
serviço em português, algo como instalação e configurações simples, pelo
menos até que eu consiga uma boa base para experimentar coisas mais
avançadas..

O que eu preciso:

Instalação prática (Debian) - consegui baixando o pacote SVN e gerando os
.debs porém não sei se devo confiar nestas configurações
Conexões com MySQL ou PostgreSQL
Integração entre dois servidores Asterisk e o OpenSIPS (roteamento de
ramais/voz)

Qualquer ajuda é bem vinda!

Abraços! :)
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Re: [OpenSIPS-Users] Callcontrol never returns duplicate callid error code

2011-09-07 Thread Saúl Ibarra Corretgé
Hi,

On Sep 7, 2011, at 3:03 PM, Dani Popa wrote:

> are you sure that is not handled as retrasmision ? Do you see the times that 
> invite hit call_control ?
> 

Assuming the INVITEs were not detected as a retransmission, there is still a 
time window in which the duplicated callID situation would not be handled 
correctly. We are working on a fix for it.


Regards,

--
Saúl Ibarra Corretgé
AG Projects




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Re: [OpenSIPS-Users] Callcontrol never returns duplicate callid error code

2011-09-07 Thread Dani Popa
are you sure that is not handled as retrasmision ? Do you see the times 
that invite hit call_control ?


dani

On 09/07/11 14:00, Mino Haluz wrote:

Hi,

I'm using kamailio+callcontrol2.0.14 , and when kamailio receives 
identical 3 INVITES, the callcontrol function never returns -3 (return 
value for duplicate callid).

What is the purpose of this return value then ?

Thanks,
M


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[OpenSIPS-Users] Callcontrol never returns duplicate callid error code

2011-09-07 Thread Mino Haluz
Hi,

I'm using kamailio+callcontrol2.0.14 , and when kamailio receives identical
3 INVITES, the callcontrol function never returns -3 (return value for
duplicate callid).
What is the purpose of this return value then ?

Thanks,
M
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Re: [OpenSIPS-Users] nat_traversal: register keepalive not, engaging

2011-09-07 Thread Gremaud Odin

Hi,


Date: Wed, 7 Sep 2011 09:44:34 +0200
From: Sa?l Ibarra Corretg?
Subject: Re: [OpenSIPS-Users] nat_traversal: register keepalive not
engaging
To: OpenSIPS users mailling list
Message-ID:<69286cac-b20e-4880-b4cc-5d89631a0...@ag-projects.com>
Content-Type: text/plain; charset=iso-8859-1

Hi,

On Sep 6, 2011, at 3:29 PM, Gremaud Odin wrote:


Hello everyone,

I began using the nat_traversal module to create a NAT traversal dedicated box, 
but I'm now stuck on something probably pretty stupid. Here is the design I'm 
working with:

Public network  ||  Private network
||
U1 --->  P1 --->  P2
||
||

U1 being the NATed UAC, P1 the NAT box and P2 an OpenSIPS Proxy/Registrar.

The registration process is OK, and I can even call another UA using U1. When checking 
the proper functioning of my script, I noticed that I could not call U1 from any other 
UA: indeed, the register keepalive did not trigger properly (no keepalive engaged for 
REGISTER, even if it pass on the function). The keepalive mechanism is working fine with 
INVITE (did not try with SUBSCRIBE), I traced OPTION messages going to U1 (which replies 
with a "404 Not Found", is this normal?). I double-checked using the module 
statistics, and there was no register keepalive, but during an dialog, it effectively 
shows that U1 is keepalived. I traced the registration process for a possible error, but 
the request is correctly transfered to P2, which responds with a 200 OK that P1 receives 
correctly. I have no more ideas about what I'm missing here...

If it can help, here is the NAT and registrar process I use:

### NAT DETECTION ###
force_rport();

# Avoid NAT detection if source IP is local
if ( client_nat_test("8")&&  $si!~"^10\.0\.0\.[0-9]{1,3}$") {
fix_contact();
append_hf("NAT-Scope: nat-relay\r\n");

# For initial invites and all subscribe
if ((is_method("INVITE")&&  !has_totag()) || is_method("SUBSCRIBE")) {
nat_keepalive();

# For registers
} else if (is_method("REGISTER")) {
nat_keepalive();
append_hf("NAT-Received: $source_uri\r\n");
}

setflag(5);
}

### REGISTRAR ###
if (is_domain_local("$fd")&&  is_method("REGISTER")) {
append_hf("Supported: Path\r\n");
add_path_received();
force_send_socket(10.0.0.1:5060);
t_relay("10.0.0.2:5060");
exit;
}

### RELAYING ###
# The request was processed by the proxy/registrar
if ($hdr(NAT-Scope)=="nat-relay"&&  !isflagset(5)) {
#$du = $hdr(NAT-Received);
$fs = $keepalive.socket($du);
# Not a subsequent message (REGISTER, SUBSCRIBE or initial INVITE), avoid 
looping on P2
} else if (!has_totag()&&  $si != "10.0.0.2") {
$du = "sip:10.0.0.2:5060";
$fs = "10.0.0.1:5060";
}

if (!t_relay("0x03")) {
sl_reply_error();
}

Any advice or idea is welcome :)


I see you call fix_contact for any type of request. Don't do this for REGISTER 
requests or you'll save the wrong received information. For a REGISTER just do 
force_rport() and nat_keepalive().

Also, the NAT test you are doing seems wrong. If an endpoint puts his own 
public IP address in the Contact header it will pass your test and no keepalive 
will be done. I usually use 3, and in the case I mentioned before the endpoint 
would fail test 2, because it would have a private address in the topmost Via.


Regards,

--
Sa?l Ibarra Corretg?
AG Projects



Thanks for the reply. I'm tired to be that dumb ;) I was looking 
everywhere, but I didn't even notice that. It solved my issue concerning 
the keepalive on register.


I know that my NAT test is not right, but this was for an 
experimentation purpose: the lab I'm using is completely local and I do 
not have any public address anywhere. So this did the trick with a NATed 
virtual machine connecting to a non NATed network. I was planning on 
using 3 also with a classic implementation.


Thanks a lot again for your reply, I can now proceed further on :)

Regards,

Odin

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[OpenSIPS-Users] a good document with examples for presence with xcap

2011-09-07 Thread Dani Popa

Hi,

I found, i think, a good document about integrating xcap with presence. 
Maybe some of you need this:


http://download.oracle.com/docs/cd/E17667_01/doc.50/e17669/cpt_concepts.htm



Dani

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Re: [OpenSIPS-Users] nat_traversal: register keepalive not engaging

2011-09-07 Thread Saúl Ibarra Corretgé
Hi,

On Sep 6, 2011, at 3:29 PM, Gremaud Odin wrote:

> Hello everyone,
> 
> I began using the nat_traversal module to create a NAT traversal dedicated 
> box, but I'm now stuck on something probably pretty stupid. Here is the 
> design I'm working with:
> 
> Public network  ||  Private network
>||
>U1 ---> P1 ---> P2
>||
>||
> 
> U1 being the NATed UAC, P1 the NAT box and P2 an OpenSIPS Proxy/Registrar.
> 
> The registration process is OK, and I can even call another UA using U1. When 
> checking the proper functioning of my script, I noticed that I could not call 
> U1 from any other UA: indeed, the register keepalive did not trigger properly 
> (no keepalive engaged for REGISTER, even if it pass on the function). The 
> keepalive mechanism is working fine with INVITE (did not try with SUBSCRIBE), 
> I traced OPTION messages going to U1 (which replies with a "404 Not Found", 
> is this normal?). I double-checked using the module statistics, and there was 
> no register keepalive, but during an dialog, it effectively shows that U1 is 
> keepalived. I traced the registration process for a possible error, but the 
> request is correctly transfered to P2, which responds with a 200 OK that P1 
> receives correctly. I have no more ideas about what I'm missing here...
> 
> If it can help, here is the NAT and registrar process I use:
> 
> ### NAT DETECTION ###
> force_rport();
> 
> # Avoid NAT detection if source IP is local
> if ( client_nat_test("8") && $si!~"^10\.0\.0\.[0-9]{1,3}$") {
>fix_contact();
>append_hf("NAT-Scope: nat-relay\r\n");
> 
># For initial invites and all subscribe
>if ((is_method("INVITE") && !has_totag()) || is_method("SUBSCRIBE")) {
>nat_keepalive();
> 
># For registers
>} else if (is_method("REGISTER")) {
>nat_keepalive();
>append_hf("NAT-Received: $source_uri\r\n");
>}
> 
>setflag(5);
> }
> 
> ### REGISTRAR ###
> if (is_domain_local("$fd") && is_method("REGISTER")) {
>append_hf("Supported: Path\r\n");
>add_path_received();
>force_send_socket(10.0.0.1:5060);
>t_relay("10.0.0.2:5060");
>exit;
> }
> 
> ### RELAYING ###
> # The request was processed by the proxy/registrar
> if ($hdr(NAT-Scope)=="nat-relay" && !isflagset(5)) {
>#$du = $hdr(NAT-Received);
>$fs = $keepalive.socket($du);
> # Not a subsequent message (REGISTER, SUBSCRIBE or initial INVITE), avoid 
> looping on P2
> } else if (!has_totag() && $si != "10.0.0.2") {
>$du = "sip:10.0.0.2:5060";
>$fs = "10.0.0.1:5060";
> }
> 
> if (!t_relay("0x03")) {
>sl_reply_error();
> }
> 
> Any advice or idea is welcome :)
> 

I see you call fix_contact for any type of request. Don't do this for REGISTER 
requests or you'll save the wrong received information. For a REGISTER just do 
force_rport() and nat_keepalive().

Also, the NAT test you are doing seems wrong. If an endpoint puts his own 
public IP address in the Contact header it will pass your test and no keepalive 
will be done. I usually use 3, and in the case I mentioned before the endpoint 
would fail test 2, because it would have a private address in the topmost Via.


Regards,

--
Saúl Ibarra Corretgé
AG Projects




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Re: [OpenSIPS-Users] nat_traversal: register keepalive not engaging

2011-09-07 Thread Gremaud Odin

Hi,

I kept on investigating this stuff, and I put openSIPS in a higher debug 
level in order to catch any suspect behavior or whatever. The only thing 
I notice in the log that seems maybe unusual is the following:


DBG:core:forward_reply: found module nat_traversal, passing reply to it
...
DBG:core:parse_params: Parsing params for:[expires=3600]
DBG:core:parse_params: Parsing params 
for:[expires=120;received="sip:1.2.3.4:12345"]

DBG:nat_traversal:get_register_expire: maximum expire for all contacts: 0

From these debug logs, I can see that the reply (the 200 OK I saw in 
the traces) is handled properly by the nat_traversal module. However, as 
the registrar function is handled by another entity, this maximum expire 
value is never set locally (it is set to 120 on the registrar entity). 
It seems that nat_traversal is not able to get a consistent expiry time, 
the one it gets being 0. Does this means that the keepalive does not 
engage, as nat_traversal believes that the expiry time is set zero ?


Please advise...

Odin


Le 06/09/2011 17:18, odin.grem...@nexcom.fr a écrit :

Date: Tue, 06 Sep 2011 15:29:18 +0200
From: Gremaud Odin
Subject: [OpenSIPS-Users] nat_traversal: register keepalive not
engaging
To: users@lists.opensips.org
Message-ID:<4e66202e.1050...@nexcom.fr>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Hello everyone,

I began using the nat_traversal module to create a NAT traversal
dedicated box, but I'm now stuck on something probably pretty stupid.
Here is the design I'm working with:

Public network  ||  Private network
  ||
  U1 --->  P1 --->  P2
  ||
  ||

U1 being the NATed UAC, P1 the NAT box and P2 an OpenSIPS Proxy/Registrar.

The registration process is OK, and I can even call another UA using U1.
When checking the proper functioning of my script, I noticed that I
could not call U1 from any other UA: indeed, the register keepalive did
not trigger properly (no keepalive engaged for REGISTER, even if it pass
on the function). The keepalive mechanism is working fine with INVITE
(did not try with SUBSCRIBE), I traced OPTION messages going to U1
(which replies with a "404 Not Found", is this normal?). I
double-checked using the module statistics, and there was no register
keepalive, but during an dialog, it effectively shows that U1 is
keepalived. I traced the registration process for a possible error, but
the request is correctly transfered to P2, which responds with a 200 OK
that P1 receives correctly. I have no more ideas about what I'm missing
here...

If it can help, here is the NAT and registrar process I use:

### NAT DETECTION ###
force_rport();

# Avoid NAT detection if source IP is local
if ( client_nat_test("8")&&  $si!~"^10\.0\.0\.[0-9]{1,3}$") {
  fix_contact();
  append_hf("NAT-Scope: nat-relay\r\n");

  # For initial invites and all subscribe
  if ((is_method("INVITE")&&  !has_totag()) || is_method("SUBSCRIBE")) {
  nat_keepalive();

  # For registers
  } else if (is_method("REGISTER")) {
  nat_keepalive();
  append_hf("NAT-Received: $source_uri\r\n");
  }

  setflag(5);
}

### REGISTRAR ###
if (is_domain_local("$fd")&&  is_method("REGISTER")) {
  append_hf("Supported: Path\r\n");
  add_path_received();
  force_send_socket(10.0.0.1:5060);
  t_relay("10.0.0.2:5060");
  exit;
}

### RELAYING ###
# The request was processed by the proxy/registrar
if ($hdr(NAT-Scope)=="nat-relay"&&  !isflagset(5)) {
  #$du = $hdr(NAT-Received);
  $fs = $keepalive.socket($du);
# Not a subsequent message (REGISTER, SUBSCRIBE or initial INVITE),
avoid looping on P2
} else if (!has_totag()&&  $si != "10.0.0.2") {
  $du = "sip:10.0.0.2:5060";
  $fs = "10.0.0.1:5060";
}

if (!t_relay("0x03")) {
  sl_reply_error();
}

Any advice or idea is welcome :)

Odin



Odin


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