date:20170120

Re: [OpenSIPS-Users] Introducing OpenSIPS 2.3

2017-01-20 Thread Bogdan-Andrei Iancu

Have you missed the "Introducing OpenSIPS 2.3" conference ? It was 
awesome, so we had it recorded for you:

 https://www.youtube.com/watch?v=4xy2vZl72Tg

Enjoy,

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

On 01/12/2017 06:39 PM, Bogdan-Andrei Iancu wrote:


A new year has arrived, so it is the time for a new OpenSIPS major 
release – forOpenSIPS version 2.3 .


For this version, the main focus on development 
is the*/“integration”/*, the integration ofOpenSIPSwith various 
external entities. Why is integration so important to end up being the 
main tag of a major release? Well, everybody in the VoIP world is 
operating VoIP platforms/systems – and these are more than SIP Engines 
(asOpenSIPSis). Indeed, the SIP Engine is the core and most important 
part of the platform, but to build something usable and useful, you 
need additional components into your platform like CDR/billing 
engines, monitoring and tracing tools, data backends, non-SIP trunking 
or more specialized SIP engines. Shortly you need your SIP Engine 
(OpenSIPS, of course) to be able to easily integrate with all these 
components.


OpenSIPS2.3  brings some new and exciting integration capabilities, 
that will definitely boost the value of your SIP platform:


  * extendedHomer/SIPCapture integration to
allow capturing of non-SIP data (transport level data, Management
Interface commands, REST queries and more);
  * SIP-I support both in terms of passing-through and in terms of
converting SIP-I to SIP and vice-versa ;
  * CGRates integration for powerful
rating/billing – everything in a simple and automatic way (via a
dedicated module);
  * FreeSWITCH  flavored Load-Balancing for a
more realistic and accurate traffic balancing over FreeSWITCH
clusters (as the load information is fetched in realtime from
FreeSWITCH);
  * theEvent Engine
to provide
support for scenarios based on Subscribe/Notify model, where the
script execution may subscribe and resume later according to
certain events (like a dynamic implementation of the Push
Notification mechanism);
  * extendedRabbitMQ  support for custom
and flexible data injection directly from OpenSIPS script;
  * extended Asynchronous support for more complex async scenarios
(like launch with no wait);
  * more end-device integration (special SIP extensions).

The timeline for OpenSIPS 2.3 is:

  * Beta Release – 13-17 March 2017
  * Stable Release – 24-28 April 2017
  * General Availability – 2nd of May 2017, duringOpenSIPS Summit


To talk more about the features of this new release, a publicaudio 
conference will be 
available on19th of January 2017, 4 pm GMT 
, thanks to 
the kind sponsorship of UberConference 
. Anyone is welcome to join to find 
out more details or to ask questions aboutOpenSIPS2.3 .


This is a public and open conference, so no registration is needed, 
but if you want to announce your intention to participate, please let 
us know via the form on the blog post 
.


Best regards,

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

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Re: [OpenSIPS-Users] Gateway failover special setup (t_check_status question)

2017-01-20 Thread Max Mühlbronner


Hi,



Actually this is very similar to what i tried before. But i just assumed 
that it is not being executed (no 408 in the trace, and no log entries 
at that moment). Now i checked again and i can see several xlog entries. 
(408 gateway timeout...) , just wasn't patient enough.



if (use_next_gw()) {
if( t_check_status("444") || !goes_to_gw("1") ) {
t_on_failure("2");
t_relay("0x08");
exit;
}

#Timeout fix #3814
if (t_check_status("408") && t_local_replied("all")) {
xlog("L_ERR", "408 gateway timeout fix test - 
rU: $rU ci: $ci");

}

...

# end of gw list / failover
t_reply("503", "Service Unavailable");
exit;



Thanks very much, i will try but this now.


BR

Max M.


On 20.01.2017 13:57, Bogdan-Andrei Iancu wrote:

Hi Max,

Something like:

 if( (t_check_status("444") || (t_check_status("408") && 
t_local_replied("all")) ) || !goes_to_gw("1") ) {


Do failover if 444 reply or if 408 without any reply received 
(internal 408).


Regards,

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

On 01/20/2017 01:21 PM, Max Mühlbronner wrote:

Hi,


my scenario is a special setup where we are checking for the reply 
code (t_check_status) and only do a failover to the next gateway when 
there is a 444 reply or gateway type is not "1".


...

if (use_next_gw()) {
if( t_check_status("444") || !goes_to_gw("1") ) {
t_on_failure("2");
t_relay();
exit;
}


The problem is, there are multiple gateways in a carrier/gatewaylist 
(gateway type 1) which are not responding (t1 timer hits and invite 
is re-transmitted) and there is no failover after the first gateway 
anymore. Is there anything i am missing, is it even possible to do 
failover only for the 444 reply while at the same time still doing a 
failover in case of Timeout-based failover ? (t1/t2 timers)



I didn't try yet, but is something like !t_check_status("\d") 
feasible? E.g. doing a failover if there is no t_check_status 
(checking regular expression for !digits?) but i guess it does not 
work that way?



if (use_next_gw()) {
if( t_check_status("444") || !goes_to_gw("1") ) {
t_on_failure("2");
t_relay();
exit;
}

if(!t_check_status("\d")) {

t_on_failure("2");
t_relay();
exit;

 }

...



BR






--
Max Mühlbronner
42com Telecommunication GmbH
Straße der Pariser Kommune 12-16
10243 Berlin
E-Mail: m...@42com.com
Web: www.42com.com

Firmenangaben/Company information:
Handelsregister/Commercial register: Amtsgericht Berlin HRB 99071 B
Umsatzsteuer-ID/VAT-ID: DE223812306
Geschäftsführer/CEO: Thomas Reinig, Alexander Reinig

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Re: [OpenSIPS-Users] [BLOG] Migrating registrations to OpenSIPS 2.2

2017-01-20 Thread Bogdan-Andrei Iancu

This issue (not being able to preserve contacts when migrating to 2.2 
version) was reported and the fix sponsored by Chris Maciejewski from 
https://voipstudio.com/ .

Thank you Chris !!

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

On 01/20/2017 02:30 PM, Ionut Ionita wrote:


Hello guys,
We just uploaded a new article on our blog about migrating your 
contacts from older versions of OpenSIPS to 2.2[0].
Starting withOpenSIPS2.2 the registered SIP contacts (stored the 
location table) have a new unique ID named *contact ID*. This new 
ID is contact specific (computed based on various contact elements) 
and it replaces the old opaque ID which was a simple DB auto-increment 
key. This creates a series of problems when you need to migrate from 
older versions and not only. You can find how to detect and solve this 
problems by reading the blog post.



[0] 
https://blog.opensips.org/2017/01/19/migrating-contacts-in-opensips-2-2/

--
Regards,
Ionut Ionita
OpenSIPS Developer


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Re: [OpenSIPS-Users] Gateway failover special setup (t_check_status question)

2017-01-20 Thread Bogdan-Andrei Iancu


Hi Max,

Something like:

 if( (t_check_status("444") || (t_check_status("408") && 
t_local_replied("all")) ) || !goes_to_gw("1") ) {


Do failover if 444 reply or if 408 without any reply received (internal 
408).


Regards,

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

On 01/20/2017 01:21 PM, Max Mühlbronner wrote:

Hi,


my scenario is a special setup where we are checking for the reply 
code (t_check_status) and only do a failover to the next gateway when 
there is a 444 reply or gateway type is not "1".


...

if (use_next_gw()) {
if( t_check_status("444") || !goes_to_gw("1") ) {
t_on_failure("2");
t_relay();
exit;
}


The problem is, there are multiple gateways in a carrier/gatewaylist 
(gateway type 1) which are not responding (t1 timer hits and invite is 
re-transmitted) and there is no failover after the first gateway 
anymore. Is there anything i am missing, is it even possible to do 
failover only for the 444 reply while at the same time still doing a 
failover in case of Timeout-based failover ? (t1/t2 timers)



I didn't try yet, but is something like !t_check_status("\d") 
feasible? E.g. doing a failover if there is no t_check_status 
(checking regular expression for !digits?) but i guess it does not 
work that way?



if (use_next_gw()) {
if( t_check_status("444") || !goes_to_gw("1") ) {
t_on_failure("2");
t_relay();
exit;
}

if(!t_check_status("\d")) {

t_on_failure("2");
t_relay();
exit;

 }

...



BR




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[OpenSIPS-Users] [BLOG] Migrating registrations to OpenSIPS 2.2

2017-01-20 Thread Ionut Ionita



Hello guys,
We just uploaded a new article on our blog about migrating your 
contacts from older versions of OpenSIPS to 2.2[0].
Starting withOpenSIPS2.2 the registered SIP contacts (stored the 
location table) have a new unique ID named *contact ID*. This new ID is 
contact specific (computed based on various contact elements) and it 
replaces the old opaque ID which was a simple DB auto-increment key. 
This creates a series of problems when you need to migrate from older 
versions and not only. You can find how to detect and solve this 
problems by reading the blog post.



[0] https://blog.opensips.org/2017/01/19/migrating-contacts-in-opensips-2-2/

--
Regards,
Ionut Ionita
OpenSIPS Developer

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Re: [OpenSIPS-Users] Opensips 1.6 as a Proxy and Presence Server

2017-01-20 Thread Carlos Oliva

you can count calls from/to a sip user using dialog profiling

You can read about it in opensips book (
https://books.google.es/books?id=y2wdDAAAQBAJ&pg=PA202&lpg=PA202&dq=opensips+dialog+count+calls&source=bl&ots=LFFoLxC9Gf&sig=fe_67u5d7xSNXm2gXUqbSOmTznM&hl=es&sa=X&ved=0ahUKEwiD6s3dy9DRAhXEUhQKHRoCAtUQ6AEIOzAD#v=onepage&q=opensips%20dialog%20count%20calls&f=false
)

To do it you create a profile with value and add the current dialog with
the sip username as value. After it you get the profile size, if it is >1
you do not do pua_setpublish because there is another ongoing call.

But as I said this problem is solved in 1.11 version, 1.6 is very old.










* _ Carlos
OlivaDepartamento de Sistemas C/ Pujades, 77-79, 8a Planta 9B | 08005
Barcelona www.numintec.com  |
carlos.ol...@numintec.com  | T: 902 02 02 97
_ Talking Numintec:
Dialogando con empresarios de éxito 
 Las soluciones en la nube de
Numintec - Casos de éxito 
 Solicita una demo

_ Medio Ambiente: Antes de
imprimir este mensaje, asegúrese de que es necesario. Nota Legal: La
información contenida en la presente transmisión es confidencial y su uso
únicamente está permitido a su(s) destinatario(s). Le informamos que los
datos personales que facilite/ha facilitado pasarán/han pasado a formar
parte de un fichero responsabilidad de NUMINTEC COMUNICACIONES S.L.. y que
tiene por finalidad gestionar las relaciones. Tiene la posibilidad de
ejercitar los derechos de acceso, rectificación, cancelación y oposición
respecto a sus datos ante la empresa, en el e-mail
comunicac...@numintec.com   o bien en el
 domicilio sito en C/ Pujades, 77-79 8ª Planta 9-B 08005 de Barcelona.*

2017-01-20 11:15 GMT+01:00 maatohewetbi :

> How did You check it? You checked calls from this sip login?
>
>
>
> --
> View this message in context: http://opensips-open-sip-
> server.1449251.n2.nabble.com/Opensips-1-6-as-a-Proxy-and-Presence-Server-
> tp7605411p7605677.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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Re: [OpenSIPS-Users] Call per second limit

2017-01-20 Thread Alain Bieuzent

Hi, 

 

You case use ratelimit module : 
http://www.opensips.org/html/docs/modules/devel/ratelimit.html, with rl_check 
function.

 

Regards

 

 

De : Users  au nom de Dragomir Haralambiev 

Répondre à : OpenSIPS users mailling list 
Date : jeudi 19 janvier 2017 à 20:25
À : OpenSIPS users mailling list 
Objet : [OpenSIPS-Users] Call per second limit

 

Hello,

 

How to made Call Per Second limitation using Opensips 2.2.2 ?

 

Regards,

Dragomir

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[OpenSIPS-Users] Gateway failover special setup (t_check_status question)

2017-01-20 Thread Max Mühlbronner


Hi,


my scenario is a special setup where we are checking for the reply code 
(t_check_status) and only do a failover to the next gateway when there 
is a 444 reply or gateway type is not "1".


...

if (use_next_gw()) {
if( t_check_status("444") || !goes_to_gw("1") ) {
t_on_failure("2");
t_relay();
exit;
}


The problem is, there are multiple gateways in a carrier/gatewaylist 
(gateway type 1) which are not responding (t1 timer hits and invite is 
re-transmitted) and there is no failover after the first gateway 
anymore. Is there anything i am missing, is it even possible to do 
failover only for the 444 reply while at the same time still doing a 
failover in case of Timeout-based failover ? (t1/t2 timers)



I didn't try yet, but is something like !t_check_status("\d") feasible? 
E.g. doing a failover if there is no t_check_status (checking regular 
expression for !digits?) but i guess it does not work that way?



if (use_next_gw()) {
if( t_check_status("444") || !goes_to_gw("1") ) {
t_on_failure("2");
t_relay();
exit;
}

if(!t_check_status("\d")) {

t_on_failure("2");
t_relay();
exit;

 }

...



BR

--
Max Mühlbronner
42com Telecommunication GmbH
Straße der Pariser Kommune 12-16
10243 Berlin
E-Mail: m...@42com.com
Web: www.42com.com

Firmenangaben/Company information:
Handelsregister/Commercial register: Amtsgericht Berlin HRB 99071 B
Umsatzsteuer-ID/VAT-ID: DE223812306
Geschäftsführer/CEO: Thomas Reinig, Alexander Reinig

Diese E-Mail enthält Informationen von 42com Telecommunication GmbH. 
Diese sind möglicherweise vertraulich und ausschließlich für den 
Adressaten bestimmt.
Sollten Sie diese elektronische Nachricht irrtümlicherweise erhalten 
haben, so informieren Sie uns bitte unverzüglich telefonisch oder per 
E-Mail.
This message is intended only for the use of the individual or entity to 
which it is addressed.

If you have received this message by mistake, please notify us immediately.

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Re: [OpenSIPS-Users] topology hiding in opensips

2017-01-20 Thread Alberto Gonzales

We used instruction in the book, which states at the end of the paragraph
this :

Topology hiding limitations
You cannot easily combine topology hiding with NAT traversal because both
the
processes mangle the Contact header. Topology hiding will not hide the
address and
other information contained in other headers such as the display in the
>From header.
To change the From header, you can use the uac_replace_from() function.

I think our problem comes from the fact that we are using nathelper and
also an rtpproxy in our script.

can anyone provide help about activating topology hiding along with
nathelper ?

thanks in advance.


On Fri, Jan 20, 2017 at 10:14 AM, Alberto Gonzales <
albertosgonz...@gmail.com> wrote:

> I forgot to mention that doing this resulted in opensips crashing after 20
> minutes :)
>
> On Fri, Jan 20, 2017 at 9:56 AM, Alberto Gonzales <
> albertosgonz...@gmail.com> wrote:
>
>> Hello grupo,
>>
>> We have configured topology hiding in opensips 2.2 this way :
>>
>> please confirm to us this is the only thing we need to do or is there
>> anything else that needs to be added.
>>
>> route {
>> 
>> 
>> if (has_totag()) {
>># sequential request withing a dialog should
>># take the path determined by record-routing
>> remplazar : if (loose_route()) {
>>if (topology_hiding_match()) {
>>
>> ...
>> ...
>>   *## esconder topologia antes de pasar la llamada*
>>topology_hiding("UC");
>>route(RELAY);
>> }
>>
>>
>> also what could be a quick test to see if this hiding is working or not.
>>
>> thanks in advance.
>>
>> Alberto
>>
>>
>
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Re: [OpenSIPS-Users] Opensips 1.6 as a Proxy and Presence Server

2017-01-20 Thread maatohewetbi

How did You check it? You checked calls from this sip login? 



--
View this message in context: 
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Sent from the OpenSIPS - Users mailing list archive at Nabble.com.

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Re: [OpenSIPS-Users] topology hiding in opensips

2017-01-20 Thread Alberto Gonzales

I forgot to mention that doing this resulted in opensips crashing after 20
minutes :)

On Fri, Jan 20, 2017 at 9:56 AM, Alberto Gonzales  wrote:

> Hello grupo,
>
> We have configured topology hiding in opensips 2.2 this way :
>
> please confirm to us this is the only thing we need to do or is there
> anything else that needs to be added.
>
> route {
> 
> 
> if (has_totag()) {
># sequential request withing a dialog should
># take the path determined by record-routing
> remplazar : if (loose_route()) {
>if (topology_hiding_match()) {
>
> ...
> ...
>   *## esconder topologia antes de pasar la llamada*
>topology_hiding("UC");
>route(RELAY);
> }
>
>
> also what could be a quick test to see if this hiding is working or not.
>
> thanks in advance.
>
> Alberto
>
>
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[OpenSIPS-Users] topology hiding in opensips

2017-01-20 Thread Alberto Gonzales

Hello grupo,

We have configured topology hiding in opensips 2.2 this way :

please confirm to us this is the only thing we need to do or is there
anything else that needs to be added.

route {


if (has_totag()) {
   # sequential request withing a dialog should
   # take the path determined by record-routing
    remplazar : if (loose_route()) {
   if (topology_hiding_match()) {

...
...
  *## esconder topologia antes de pasar la llamada*
   topology_hiding("UC");
   route(RELAY);
}


also what could be a quick test to see if this hiding is working or not.

thanks in advance.

Alberto
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Re: [OpenSIPS-Users] Opensips 1.6 as a Proxy and Presence Server

2017-01-20 Thread Carlos Oliva

Hi Marcin:

I had the same problem years ago with 1.6 version. In the first call BLF
works but the second call fails. I solved it using dialog module to check
the mumber of calls and not doing pua_set_publish if the mubers of calls
are > 1

I think this problem was solved on 1.11 version, wich is suported until 7th
of May 2017, 1.6 version has reached end of life on 2010

regards,










* _ Carlos
OlivaDepartamento de Sistemas C/ Pujades, 77-79, 8a Planta 9B | 08005
Barcelona www.numintec.com  |
carlos.ol...@numintec.com  | T: 902 02 02 97
_ Talking Numintec:
Dialogando con empresarios de éxito 
 Las soluciones en la nube de
Numintec - Casos de éxito 
 Solicita una demo

_ Medio Ambiente: Antes de
imprimir este mensaje, asegúrese de que es necesario. Nota Legal: La
información contenida en la presente transmisión es confidencial y su uso
únicamente está permitido a su(s) destinatario(s). Le informamos que los
datos personales que facilite/ha facilitado pasarán/han pasado a formar
parte de un fichero responsabilidad de NUMINTEC COMUNICACIONES S.L.. y que
tiene por finalidad gestionar las relaciones. Tiene la posibilidad de
ejercitar los derechos de acceso, rectificación, cancelación y oposición
respecto a sus datos ante la empresa, en el e-mail
comunicac...@numintec.com   o bien en el
 domicilio sito en C/ Pujades, 77-79 8ª Planta 9-B 08005 de Barcelona.*

2017-01-20 8:58 GMT+01:00 maatohewetbi :

> I think I found the reason BLF doesn't work. I made a test. I've erased
> presenity table. Then:
> 1. I called other user, led is blinking, and presentity table shows record
> with xml bodies with early state.
> 2. Then I pickup a call, the same record changes with confirmed state.
> 3. Then I end this call, the same record changes to terminated state.
>
> So first call was ok. But after it I made second call, but it turned out
> that presentity table wasn't erased, and previous records exist. So when I
> make this second call, xml body contains previous xml, and this right one!
> I
> think presentity table should be erased after a call. Do You know the
> reason?
>
>
>
>
> --
> View this message in context: http://opensips-open-sip-
> server.1449251.n2.nabble.com/Opensips-1-6-as-a-Proxy-and-Presence-Server-
> tp7605411p7605672.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
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Re: [OpenSIPS-Users] Introducing OpenSIPS 2.3

Re: [OpenSIPS-Users] Gateway failover special setup (t_check_status question)

Re: [OpenSIPS-Users] [BLOG] Migrating registrations to OpenSIPS 2.2

Re: [OpenSIPS-Users] Gateway failover special setup (t_check_status question)

[OpenSIPS-Users] [BLOG] Migrating registrations to OpenSIPS 2.2

Re: [OpenSIPS-Users] Opensips 1.6 as a Proxy and Presence Server

Re: [OpenSIPS-Users] Call per second limit

[OpenSIPS-Users] Gateway failover special setup (t_check_status question)

Re: [OpenSIPS-Users] topology hiding in opensips

Re: [OpenSIPS-Users] Opensips 1.6 as a Proxy and Presence Server

Re: [OpenSIPS-Users] topology hiding in opensips

[OpenSIPS-Users] topology hiding in opensips

Re: [OpenSIPS-Users] Opensips 1.6 as a Proxy and Presence Server

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