Re: [OpenSIPS-Users] New OpenXCAP release 2.4.0

2016-03-03 Thread Adrian Georgescu
On Debian and Ubuntu you can just do apt-get install openxcap from AGP repos. 
If you build related dependencies from scratch you are here not in the right 
place as this has nothing to do with OpenSIPS or OpenXCAP themselves.

Adrian

> On 28 Feb 2016, at 23:15, sevpal  wrote:
> 
> Hi, there is a message "No Module named sipsimple.core" when tried to start 
> the OpenXCAP server. How can this be resolved? I'm thinking it is referring 
> to the "python-sipsimple" package but that is a beast to compile along with 
> ffmpeg.
> 
> 
> -Original Message- From: Adrian Georgescu
> Sent: Thursday, January 07, 2016 1:06 PM
> To: OpenSIPS users mailling list ; OpenSIPS devel mailling list
> Subject: [OpenSIPS-Users] New OpenXCAP release 2.4.0
> 
> Hello,
> 
> There is a new release of OpenXCAP with various bug fixes
> 
> openxcap (2.4.0) unstable; urgency=medium
> 
> * Removed runtime dependency check
> * Updated copyright years
> 
> openxcap (2.3.0) unstable; urgency=medium
> 
> * Refactor xcapdiff publishing for OpenSIPS backend
> * Fix dependency name
> * Drop dependency on twisted-web
> * Raise python-application version dependency
> * Add dependency on python-sipsimple
> 
> http://openxcap.org
> 
> Regards,
> Adrian
> 
> 
> 
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Re: [OpenSIPS-Users] Solaris Sparc Compile Error

2016-03-03 Thread Nathaniel L. Keeling III

Razvan,

Where can I open this request at?

Thanks

Nathaniel

On 3/2/16 4:53 AM, Răzvan Crainea wrote:

Hi, Nathaniel!

Unfortunately there is no way to specify the GNU utilities. Please 
open a pull request for this, I will try to take care of it asap.


Best regards,
Răzvan

On 03/01/2016 08:33 AM, Nathaniel L. Keeling III wrote:

Razvan,

When compiling version 1.8, it was not detecting the 64 bit properly 
which is why I added the variables. I will try building without any 
options set. As far as "sed" is concern, I have gsed version 4.2.2 
installed which is what I thought was being used. It sounds like the 
compile is pulling in the original "sed" that came with the OS. Here 
is the version for that Solaris sed: VERSION:  
11.10.0,REV=2005.01.21.15.53


Is there a way to specify/utilize the GNU utilities?

Thanks

Nathaniel


On 2/29/16 8:54 AM, Răzvan Crainea wrote:

Hi, Nathaniel!

In the fist pastebin, are you sure you need the -m64 options? The 
makefile system should detect the architecture and build the packets 
accordingly. Anyway, this is just a question, the real problem seems 
to be the version of the sed package. Can you tell me what sed 
version are you using? Perhaps pasting a help command, since I don't 
have access to a Solaris macine.


All the other pastebin logs do not use the latest version, that's 
why they do not compile.


Best regards,
Răzvan


On 02/29/2016 02:13 PM, Nathaniel L. Keeling III wrote:

Razvan,

Yes, setting LD_EXTRA_OPTS made it compile. I did a git pull and 
tried a few things and here are the results:


I removed the env variables CFLAGS and CXXFLAGS and then set env 
variables CC_EXTRA_OPTS=-m64 and LD_EXTRA_OPTS=-m64. I then 
executed the compile from "make menuconfig" and here is the results.


http://pastebin.com/VwtLcDqw


I then reran the compile with env CFLAGS=-m64 and CXXFLAGS=-m64 
using this:


CC_EXTRA_OPTS=-m64 LD_EXTRA_OPTS=-m64 NICER=0 make all

Here is the pastebin link for that output:

http://pastebin.com/2mpXuehv


Here is the results of the compile with the env CFLAGS and CXXFLAG 
not set using:

CC_EXTRA_OPTS=-m64 LD_EXTRA_OPTS=-m64 NICER=0 make all

This is the pastebin link:

http://pastebin.com/0TW4ZU4S


I then tried again with the env variables CFLAGS and CXXFLAG set as 
before with this:

LD_EXTRA_OPTS=-m64 NICER=0 make all

Here is the pastebin link:

http://pastebin.com/emUFTGu1



Thanks

Nathaniel

On 2/29/16 4:52 AM, Răzvan Crainea wrote:

LD_EXTRA_OPTS=-m64 NICER=0 make all




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Re: [OpenSIPS-Users] mhome asymmetric port issue

2016-03-03 Thread Satish Patel
Thanks Razvan,

Problem is we have legacy application running on ser-0.10 older
version and we have did lots of other customization in ser I tried to
use force_send_socket() but look like that support isn't in RR module.

if i enable "mhomed=1" and use t_relay() in dispatcher then its
consuming REGISTER packet and sending AUTH challenge to client instead
of sending that REGISTER to backend dispatcher..

if i use forward(uri:host, uri:port); function then it doesn't
understand socket correctly. atleast t_relay() is working but
consuming REGISTER, we have very simple code like following

else if ( (method=="REGISTER")  || (method=="INVITE") ) {
if ( !ds_select_dst("2", "2") ) {
xlog("L_ERR", "Unable to route REGISTER\n");
sl_send_reply("500","Unable to route REGISTER");
break;
}
..
..
t_relay()

On Thu, Mar 3, 2016 at 3:39 AM, Răzvan Crainea  wrote:
> Hi, Satish!
>
> By default, OpenSIPS uses the same interface to send the reply. However,
> when using mhomed=1, the operating system decides where the reply should be
> sent to. And in your case, the operating system simply chooses a different
> interface. So it seems this is the normal behavior, there's nothing wrong.
> If you really want to use the same interface for replies, you should use the
> force_send_socket() function to set the desired interface.
>
> Best regards,
> Răzvan
>
>
> On 03/02/2016 11:10 PM, Satish Patel wrote:
>>
>> mhome=1
>> listen=udp:10.0.0.1:6060 udp:10.0.0.1:5060 udp:192.168.100.1:6060
>> udp:192.168.100.1:5060
>>
>>  From client when i send REGISTER to 5060 then server sending reply
>> back using port 6060, it should send reply back client using 5060
>> right???
>>
>> If i use mhome=0 everything works!
>>
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[OpenSIPS-Users] How to know that caller and callee talk from differente networks?

2016-03-03 Thread Rodrigo Pimenta Carvalho

Hi.


In my script OpenSIPS.cfg sometimes I do need to call "set_advertised_address" 
and sometimes I don't.


Basically, I want to call this function only for messages between caller and 
callee that is registered from different networks. Ex: caller is in the same 
network (LAN) as the sip proxy and the callee is somewhere on Internet.


So, how to know that, during a dialog, caller and callee are in 2 different 
networks?


When they are in different networks, I want to call set_advertised_address for 
SIP OK and SIP BYE messages.


Any hint will be very helpful!


Thanks a lot.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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Re: [OpenSIPS-Users] Bflag problem!

2016-03-03 Thread Dragomir Haralambiev
Hi,

I make test. All working fine now!

Thanks,
Dragomir

2016-03-03 17:38 GMT+02:00 Liviu Chircu :

> Hello Dragomir,
>
> Thank you for the report - I just pushed a fix for this on the master
> branch!
>
> Liviu Chircu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 03.03.2016 16:41, Dragomir Haralambiev wrote:
>
> Hello,
>
> After latest update of Opensips 2.2 I have problem with "setbflag".
> I can not set bflag.
> Here is example:
>
> 
> modparam("nathelper", "remove_on_timeout_bflag", "RM_ONTO_FLAG")
> .
>
> setbflag(RM_ONTO_FLAG);
> if (isbflagset(RM_ONTO_FLAG)) {
> xlog("L_ERR", "RM_ONTO_FLAG is set");
>
> }
>
> Best regards,
> Dragomir
>
>
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Re: [OpenSIPS-Users] Bflag problem!

2016-03-03 Thread sevpal
I noticed that too with 2.1 latest git version, branch flags are not being 
saved in the location table. I reverted as a result. This is just to confirm. 

From: Dragomir Haralambiev 
Sent: Thursday, March 03, 2016 9:41 AM
To: OpenSIPS users mailling list 
Subject: [OpenSIPS-Users] Bflag problem!

Hello, 

After latest update of Opensips 2.2 I have problem with "setbflag".
I can not set bflag.
Here is example:


modparam("nathelper", "remove_on_timeout_bflag", "RM_ONTO_FLAG")

.

setbflag(RM_ONTO_FLAG); 
if (isbflagset(RM_ONTO_FLAG)) {
xlog("L_ERR", "RM_ONTO_FLAG is set");

}

Best regards,
Dragomir



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[OpenSIPS-Users] Rate Limit Module

2016-03-03 Thread Richard Robson
I'm using 2.1 and cant see the ratelimit module limiting calls. I'm just
testing at the moment but i;'ve set the limit to 2 and i'm making 2
simultaneous calls from the downstream asterisk. tHis is then
loadbalancing to out upstream sip supplier

This is the part catching the INVITE then loadbalancing and rate
limiting. The loadbalancer is working OK.

if (is_method("INVITE")) {

if (lb_start("2","calls")) {

setflag(SST_FLAG);
create_dialog('B');
$var(limit) = 2;
if (!rl_check("$du", "$var(limit)", "RED")) { #
Check Rate limit


if (lb_start("3","calls")) {

setflag(SST_FLAG);
create_dialog('B');
$var(limit) = 1;
if (!rl_check("$du",
"$var(limit)","RED")) {
sl_send_reply("503",
"Server Unavailable");

exit;
}
}
}
}

t_on_branch("per_branch_ops");
t_on_reply("handle_nat");
t_on_failure("missed_call");
 
if (rtpengine_offer())

}


If I make 3 calls I would expect one of the calls to get dropped and to
use the other load balancer pool I can see the Counter going to 3 which
is more than the limit of 2 and the drop rate at 0. I am getting the
same results using TAILDROP too.

[root@gl-sip-01 ~]# opensipsctl fifo rl_list
PIPE::  id=sip:sip-wholesale.aql.com algorithm=RED limit=2 counter=3
PIPE::  drop_rate=0
[root@gl-sip-01 ~]# opensipsctl fifo rl_list
PIPE::  id=sip:sip-wholesale.aql.com algorithm=RED limit=2 counter=3
PIPE::  drop_rate=0
[root@gl-sip-01 ~]# opensipsctl fifo rl_list
PIPE::  id=sip:sip-wholesale.aql.com algorithm=RED limit=2 counter=3
PIPE::  drop_rate=0
[root@gl-sip-01 ~]# opensipsctl fifo rl_list
PIPE::  id=sip:sip-wholesale.aql.com algorithm=RED limit=2 counter=0
PIPE::  drop_rate=0
[root@gl-sip-01 ~]# opensipsctl fifo rl_list
PIPE::  id=sip:sip-wholesale.aql.com algorithm=RED limit=2 counter=0
PIPE::  drop_rate=0
[root@gl-sip-01 ~]# opensipsctl fifo rl_list
PIPE::  id=sip:sip-wholesale.aql.com algorithm=RED limit=2 counter=0
PIPE::  drop_rate=0

Regards,

-- 
Richard Robson
Greenlight Support


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[OpenSIPS-Users] Nathelper module doesn't send OPTIONS keepalive

2016-03-03 Thread Julian Santer

Hi guys,

we tried to switch from nat_traversal to nathelper.
The reason is the keepalive mechanism.

The nat_traversal module sends OPTIONS with the following to header: 
sip:UAC_IP:UAC_PORT
Most of the UAC's answers with a 404 Not found.
On AVM Fritzbox with firmware >= 6.04, this OPTIONS may activate a security 
feature.
So after a certain time, the Fritzbox blocks all packages send from our proxy.
As we have ca. 80% AVM Fritzbox as UAC, we got a big problem.
So we deactivated the nat_keepalive vor this UAC's and we have to enable the 
keepalive Feature on the Fritzbox.

The better solution would be, if we could send OPTIONS with a to header like: 
sip:username@UAC_IP:UAC_PORT.
As I understood the nathelper module could send OPTIONS like this. Because it 
is looking into the userloc table. Right?

The nathelper module is on our edge server, the registrar on our core server.
The problem now is, that the module does not sends any OPTIONS at all.

We are using the version 2.1.2 from git. Last pull on 2016/03/01.

The nathelper config looks like:
 nathelper module
loadmodule "nathelper.so"
modparam("nathelper", "natping_interval", 56)
modparam("nathelper", "ping_nated_only", 0)
modparam("nathelper", "natping_partitions", 1)
modparam("nathelper", "natping_socket", "IP_PROXY:5060")
modparam("nathelper", "sipping_bflag", "SIP_PING_FLAG")
modparam("nathelper", "sipping_from", "sip:keepalive@DEFAULT_REALM")
modparam("nathelper", "sipping_method", "OPTIONS")
modparam("nathelper", "nortpproxy_str", "")
modparam("nathelper", "natping_tcp", 0)

For the test, in the register route I tried to set always the sipping_bflag:
if (proto == UDP)
{
setbflag(SIP_PING_FLAG);
xlog("L_INFO", "Nat keepalive sip_ping_flag - LF_BASE");
}

But in my traces I can't find any OPTIONS send by the nathelper module.
Could you give me a hint?

Kind regards,
Julian Santer
Raiffeisen OnLine


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Re: [OpenSIPS-Users] Bflag problem!

2016-03-03 Thread Liviu Chircu

Hello Dragomir,

Thank you for the report - I just pushed a fix for this on the master 
branch!


Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 03.03.2016 16:41, Dragomir Haralambiev wrote:

Hello,

After latest update of Opensips 2.2 I have problem with "setbflag".
I can not set bflag.
Here is example:


modparam("nathelper", "remove_on_timeout_bflag", "RM_ONTO_FLAG")
.

setbflag(RM_ONTO_FLAG);
if (isbflagset(RM_ONTO_FLAG)) {
xlog("L_ERR","RM_ONTO_FLAG is set");

}
Best regards,
Dragomir


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[OpenSIPS-Users] Bflag problem!

2016-03-03 Thread Dragomir Haralambiev
Hello,

After latest update of Opensips 2.2 I have problem with "setbflag".
I can not set bflag.
Here is example:


modparam("nathelper", "remove_on_timeout_bflag", "RM_ONTO_FLAG")
.

setbflag(RM_ONTO_FLAG);
if (isbflagset(RM_ONTO_FLAG)) {
xlog("L_ERR", "RM_ONTO_FLAG is set");

}

Best regards,
Dragomir
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Re: [OpenSIPS-Users] Fwd: Push notifications

2016-03-03 Thread Babil (Golam Sarwar)
Hi José,
thanks for the details. I have Grocer gem with Rails.

> Do you send multiple INVITEs to the SIP server until Bob becomes online?

Yes. If Alice receives 404 when trying to reach Bob, we retry for a given
number of seconds, until Bob registers and can accept the call.


On 02-Mar-16 5:59 AM, José Expósito wrote:
> Hi Babil,
>
> I use the rest_post() method to send a query to my server:
>
> > rest_post("$var(pns_push_url)", "{ \"user\": { \"uri\":\"$var(uri)\",
> \"type\":\"INCOMING_CALL\" } }", "application/json", "$var(ct)", 
> "$var(rcode)");
>
> You can use async with rest_post if it takes too long.
> The server is just a Ruby on Rails server that uses the houston gem to send
> the notifications: https://github.com/nomad/houston
>
> About having Alice retying... How do you do so? Do you send multiple INVITEs
> to the SIP server until Bob becomes online?
>
> Jose
>
> 2016-03-02 14:48 GMT+01:00 Babil (Golam Sarwar)  >:
>
> Hi José,
> thank you. I have a somewhat similar implementation but on top of OpenSIPS
> 1.11.
>
> I run a daemon that waits for signals from OpenSIPS for triggering push
> notifications. Then the rest of the REST-API call (implementing the APN
> functionality) is handled within that separate daemon's thread. OpenSIPS 
> is
> blocked for minimum amount of time during the REST call. How do you make 
> your
> REST calls from the OpenSIPS script? Can you add an example?
>
> Also, in order to wake up Bob and receive calls, in my system I have 
> basically
> made Alice to retry a given number of times. Since in my scenario, I 
> control
> my SIP clients, it is feasible. But this may not apply to your case.
>
>
> On 3/2/16 5:13 AM, José Expósito wrote:
> > Hi Babil,
> >
> > Thank you very much for your answer.
> >
> > I have another server that offers a RESTful API to send the push 
> notifications,
> > store the tokens, etc.
> >
> > I'm using OpenSIPS *2.1.2*
> >
> > Jose
> >
> > 2016-03-02 14:00 GMT+01:00 Babil (Golam Sarwar)  
> > >>:
> >
> > Hi José,
> > what is your mechanism to initiate sending the push? For example, 
> what
> > function/script are you calling? Please also add the OpenSIPS
> version you are
> > using.
> >
> > On 3/2/16 4:47 AM, José Expósito wrote:
> > > Hi all,
> > >
> > > I'm trying to implement Apple Push Notifications with OpenSIPS.
> > >
> > > At the moment, when Alice calls Bob and Bob is offline, I send the
> push
> > > notification to Bob and call async sleep until Bob becomes online
> and I
> > proceed
> > > whit the call as usual:
> > >
> > > > if (!lookup("location","m")) {
> > > > if (is_method("INVITE")) {
> > > > # Sends the notification here
> > > > t_reply("100", "Trying");
> > > > t_reply("180", "Ringing");
> > > > async(sleep("5"), wait_bob);
> > > > }
> > > >
> > > > [...]
> > > >
> > > > route[wait_bob] {
> > > > if (!lookup("location","m")) {
> > > > async(sleep("2"), wait_bob);
> > > > }
> > > > if (!t_relay()) { # This establishes the call as usual
> when Bob
> > > becomes online
> > > > send_reply("500","Internal Error");
> > > > }
> > > >exit;
> > > > }
> > >
> > >
> > > This mechanism works without problems, but, when integrating and
> RTP proxy
> > > OpenSIPS is randomly crashing after a while.
> > >
> > > That's why I'm trying to avoid the use of async sleep and switch
> to another
> > > mechanism.
> > >
> > > As usual, how do you solve this problem?
> > > Is there any  way to send a new (re)INVITE when Bob becomes 
> online?
> > >
> > > I'm trying to use create_dialog() when Bob registers, but it
> doesn't look to
> > > work -I'm probably using it wrong.
> > >
> > > Thank you very much in advance
>

--
Regards,
Babil (Golam Sarwar)

PGP Key Download URL: http://bit.ly/gsbabil-pgp-key
PGP Key Fingerprint : D3A1 EED0 5BA0 72D3 A011 75CB 8EA6 7D99 F433 E92D

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Re: [OpenSIPS-Users] send error message to specific server.

2016-03-03 Thread guillaume Montassier

Hi Alain,

Our case:
When an INVITE arrive to our main server, he send the invite to a 
succession of servers (Theses servers talk in stateless mode via cmd 
"send").
When all treatements are done, one of theses stateless servers "send" 
the INVITE to the main server. This way, all the conversation between 
stateless servers are invisible.
This is working, but when we raise an error (like 404) we didn't suceed 
to send it directly to the main server.


Thanks,
Guillaume

On 02/03/2016 17:45, Alain Bieuzent wrote:

Hi Guillaume,

in which case ? what is the trigger for sending this 404?

Regards

Le 02/03/2016 09:29, guillaume Montassier a écrit :

Hi,

Is it possible to send a SIP error (eg. 404 Not found) to a specific
server?
Like "sl_send_reply("404", "Not found")" but not as a reply, as a new
message send to a specific server.

Thanks,
Guillaume

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Re: [OpenSIPS-Users] mhome asymmetric port issue

2016-03-03 Thread Răzvan Crainea

Hi, Satish!

By default, OpenSIPS uses the same interface to send the reply. However, 
when using mhomed=1, the operating system decides where the reply should 
be sent to. And in your case, the operating system simply chooses a 
different interface. So it seems this is the normal behavior, there's 
nothing wrong. If you really want to use the same interface for replies, 
you should use the force_send_socket() function to set the desired 
interface.


Best regards,
Răzvan

On 03/02/2016 11:10 PM, Satish Patel wrote:

mhome=1
listen=udp:10.0.0.1:6060 udp:10.0.0.1:5060 udp:192.168.100.1:6060
udp:192.168.100.1:5060

 From client when i send REGISTER to 5060 then server sending reply
back using port 6060, it should send reply back client using 5060
right???

If i use mhome=0 everything works!

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