Re: [OpenSIPS-Users] No TLS-related files in OpenSIPS 1.11.9 src.tar.gz

[Răzvan Crainea]

Hi, Chen-Che!

Are you talking about the deb files generated here [1]?
If not, can you send me a link to the affected archive?

[1] http://apt.opensips.org/packages.php

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 11/08/2016 04:01 AM, Chen-Che Huang wrote:

Hi Răzvan,

Thanks for your help. The updated archive contains the TLS-related files.
However, the generated deb files from the archived are having the 1.11.8-1
on their names rather than 1.11.9. Can you kindly help to take a look?
Thanks.

Best regards,
Chen-Che



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[OpenSIPS-Users] Regarding Option request .

[Sasmita Panda]

Hi All ,

I am using opensips-1.11 . I have a requirement that when an
"Option" request will come to my proxy from the gateways I have added in my
dr_gateways table then only I will process the request and will send 200 OK
.

 Previously , I was sending 200 Ok to all the Options requests , So I
have written the config file like bellow .

route{
 if (method=="OPTIONS") {
sl_send_reply("200", "OK");
exit;
}
...
...
route(1);
}

I have tried many possible ways but its not happening . What should I do to
achieve my goal ?

*Thanks & Regards*
*Sasmita Panda*
*Network Testing and Software Engineer*
*3CLogic , ph:07827611765*
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Re: [OpenSIPS-Users] Regarding Option request .

[Răzvan Crainea]

Hello!

You chould use the is_from_gw()[1] function to detect if a message is 
coming from a gateway. So basically your code should look like this:


if (method == "OPTIONS" && is_from_gw()) {
sl_send_reply("200", "OK");
exit;
}

[1] http://www.opensips.org/html/docs/modules/1.11.x/drouting.html#id295343

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 11/08/2016 10:05 AM, Sasmita Panda wrote:

Hi All ,

I am using opensips-1.11 . I have a requirement that when an 
"Option" request will come to my proxy from the gateways I have added 
in my dr_gateways table then only I will process the request and will 
send 200 OK .


 Previously , I was sending 200 Ok to all the Options requests , 
So I have written the config file like bellow .


route{
 if (method=="OPTIONS") {
sl_send_reply("200", "OK");
exit;
}
...
...
route(1);
}

I have tried many possible ways but its not happening . What should I 
do to achieve my goal ?


*/Thanks & Regards/*
/Sasmita Panda/
/Network Testing and Software Engineer/
/3CLogic , ph:07827611765/


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Re: [OpenSIPS-Users] B2B top hiding

[Denis]

Hello, Razvan!

I found the problem (that was not correctly analyzing log).
Everything working if i set host and port before b2b_init_request. 

Thank you for help.

 mailto:denis7...@mail.ru


Hi, Denis!

Are you seeing any other ERROR messages in the logs? When you see those errors 
in the log, does OpenSIPS send any message out (even to himself)? Can you 
provide a full trace?

Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 11/07/2016 03:20 PM, Denis wrote:

Re: [OpenSIPS-Users] B2B top hiding Hello, Razvan!

I want to make some input point to my SIP network which will make 
authentication and topology hiding.
Authentication works (i checked it by inserting in client SIP UA wrong 
password), but b2b doesn`t. 
When i try to setting port before b2b_init_request i see in log " 
ERROR:b2b_logic:create_top_hiding_entities: failed to create new b2b server 
instance"

mailto:denis7...@mail.ru


Hi, Denis!

Can you please detail a bit what you are trying to achieve? In the trace 
attached, all I can see is that OpenSIPS doesn't authenticate the UA - I don't 
see why it should even try to send the INVITE anywhere.
However, if you are saying that you see both lines in the log, it means at a 
certain point the INVITE was authenticated and the topo-hiding was engaged. Did 
you try setting the port before calling b2b_init_request(), in route(1)?

BR
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 11/07/2016 02:20 PM, Denis wrote:

B2B top hiding Hello!

Opensips 2.2.2.

I want to make an installation of Opensips which  will include auth and top 
hiding functionality.

First of all, in the main route, i catch INVITE
"if  (is_method("INVITE") && !has_totag()) {
route(1);
exit;
}"

then, in route [1], i make some auth procedure using "pv_proxy_authorize" and 
"proxy_authorize" functions
if auth procedure is successful i make 
"xlog("L_INFO", "Route1:$rm was received with auth (IP=$si, IPD=$rd, 
CALLID=$ci, FROMTAG=$ft, TOTAG=$tt, AUTH=$au) AND B2B prepared");
b2b_init_request("top hiding");
exit;"

In the local route i try to edit port of the destination, using 
"xlog("L_INFO", "Local:$rm was received with auth (IP=$si, IPD=$rd, CALLID=$ci, 
FROMTAG=$ft, TOTAG=$tt, AUTH=$au)");
rewritehostport("1.1.1.1:5065");" 

where 1.1.1.1 - ip address of Opensips.

In attachment you can see debug of unsuccessful call, where 
1.1.1.1 - ip address of Opensips.
2.2.2.2 - caller SIP UA

I can see in syslog both messages, in local route and in route [1].

As you can see from the debug Opensips doesn`t try to send INVITE to 
1.1.1.1:5065.

Thank you for any help.


mailto:denis7...@mail.ru 

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[OpenSIPS-Users] Common points in B2B top hiding

[Denis]

Hello!

Sorry that it was, but are there any common points between received INVITE and 
new generated INVITE while using B2B top hiding?

I need some points, gotten in local_route, which will uniquely identified the 
call and which i can use, for example, to add some headers (based on that 
points).

Thank you for any help.



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[OpenSIPS-Users] Pickup group help

[Schneur Rosenberg]

Can someone please guide me how I should build pickup groups feature in
OpenSIPS?

thank you
S. Rosenberg
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Re: [OpenSIPS-Users] Common points in B2B top hiding

[Răzvan Crainea]

Hi, Denis!

You can copy headers from the initial request to the B2B one by using 
the custom headers[1].


[1] http://www.opensips.org/html/docs/modules/2.2.x/b2b_logic.html#id293556

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 11/08/2016 03:41 PM, Denis wrote:

Common points in B2B top hiding Hello!

Sorry that it was, but are there any common points between received 
INVITE and new generated INVITE while using B2B top hiding?


I need some points, gotten in local_route, which will uniquely 
identified the call and which i can use, for example, to add some 
headers (based on that points).


Thank you for any help.



mailto:denis7...@mail.ru


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Re: [OpenSIPS-Users] Common points in B2B top hiding

[Denis]

Hello, Razvan!

No, it is not quite what i am looking for.

For example,  before top hiding i make authentication procedure during which i 
got login.
For some call processing logging i need to translate this login to some proxy 
instance.

How can i do that?

mailto:denis7...@mail.ru


Hi, Denis!

You can copy headers from the initial request to the B2B one by using the 
custom headers[1].

[1] http://www.opensips.org/html/docs/modules/2.2.x/b2b_logic.html#id293556

Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 11/08/2016 03:41 PM, Denis wrote:

Common points in B2B top hiding Hello!

Sorry that it was, but are there any common points between received INVITE and 
new generated INVITE while using B2B top hiding?

I need some points, gotten in local_route, which will uniquely identified the 
call and which i can use, for example, to add some headers (based on that 
points).

Thank you for any help.



mailto:denis7...@mail.ru 

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[OpenSIPS-Users] How to find out which TCP sockes OpenSIPS is listening on?

[Rodrigo Pimenta Carvalho]

Hi.


Is it possible to find which is every socket that is currently opened to 
opensips listens on SIP messages from peers, while using TCP?


I have examined opensipsctl command, but it doesn't show the sockets.

I need see if a new socket is being created and opened when a peer sends a SIP 
UPDATE.


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] Common points in B2B top hiding

[Răzvan Crainea]

Hi, Denis!

Take a look at the "a" flag used by the b2b_init_request()[1] function. 
This might help you achieve what you want.
May I ask you why you are using the b2b topology hiding, and not the 
dedicated module for topology hiding[2]? It would be far more easier to 
use the latter one in most of the cases.


[1] http://www.opensips.org/html/docs/modules/2.2.x/b2b_logic.html#id294010
[2] http://www.opensips.org/html/docs/modules/2.2.x/topology_hiding.html

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 11/08/2016 04:19 PM, Denis wrote:

Re: [OpenSIPS-Users] Common points in B2B top hiding Hello, Razvan!

No, it is not quite what i am looking for.

For example,  before top hiding i make authentication procedure during 
which i got login.
For some call processing logging i need to translate this login to 
some proxy instance.


How can i do that?

mailto:denis7...@mail.ru


Hi, Denis!

You can copy headers from the initial request to the B2B one by using 
the custom headers[1].


[1] 
http://www.opensips.org/html/docs/modules/2.2.x/b2b_logic.html#id293556


Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com 
On 11/08/2016 03:41 PM, Denis wrote:

Common points in B2B top hiding Hello!

Sorry that it was, but are there any common points between received 
INVITE and new generated INVITE while using B2B top hiding?


I need some points, gotten in local_route, which will uniquely 
identified the call and which i can use, for example, to add some 
headers (based on that points).


Thank you for any help.



mailto:denis7...@mail.ru

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Re: [OpenSIPS-Users] How to find out which TCP sockes OpenSIPS is listening on?

[Răzvan Crainea]

Hi, Rodrigo!

So are you interested in finding the opened TCP connections? If so, you 
should try the list_tcp_conns command[1].


[1] http://www.opensips.org/Documentation/Interface-CoreMI-2-2#toc5

Best regards,

Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 11/08/2016 06:10 PM, Rodrigo Pimenta Carvalho wrote:



Hi.


Is it possible to find which is every socket that is currently opened 
to opensips listens on SIP messages from peers, while using TCP?



I have examined opensipsctl command, but it doesn't show the sockets.

I need see if a new socket is being created and opened when a peer 
sends a SIP UPDATE.



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] Support for https in async

[Ramachandran, Agalya (Contractor)]

Hi Liviu,

Thank you for response. I have few questions, can you please clarify me on 
these.


* Is there is a roadmap in future targeting async() to handle https 
queries?

* In start_async_http_request(), why is curl_multi_wait is not used to 
overcome 1024 fds limitation? and is there plan to use it in future?

* What is the use of read_fds[] in rest_methods.c and who uses it?

* Why select() is not used ?


Regards,
Agalya
From: Liviu Chircu [mailto:li...@opensips.org]
Sent: Monday, October 31, 2016 12:27 PM
To: Ramachandran, Agalya (Contractor) ; 
OpenSIPS users mailling list 
Subject: Re: Support for https in async


Hi, Agalya!

We have not done any development in that direction, and I assume it won't work 
out of the box, as we need to instruct libcurl on where to find the CA 
certificate before we can expect it to establish the TLS connection. As an 
alternative, we could add the option of disabling host/peer verifications like 
here [1]

[1]: https://curl.haxx.se/libcurl/c/https.html

Liviu Chircu

OpenSIPS Developer

http://www.opensips-solutions.com
On 31.10.2016 16:41, Ramachandran, Agalya (Contractor) wrote:
Hi team,

Have you tried with https://url with async and does that work without issue ?
When I try, resume_async_http_req is called, but crash is occurring at libcurl 
library. So helpless to debug further.
async(rest_put("https://url";, "$fU,$tU,$ci ", "application/json", 
"$var(body)", "$var(ct)", "$var(rcode)"),resume);

My question is crash occring only in my scenario or  OpenSIPS doesn't support 
async as https?
Here is my dump just in case for reference.

#0  0x7f567248da1e in Curl_raw_nequal () from /lib64/libcurl.so.4
#1  0x7f567245bd0f in Curl_checkheaders () from /lib64/libcurl.so.4
#2  0x7f567245d1e5 in Curl_http () from /lib64/libcurl.so.4
#3  0x7f567246db4b in Curl_do () from /lib64/libcurl.so.4
#4  0x7f567247da1b in multi_runsingle () from /lib64/libcurl.so.4
#5  0x7f567247e121 in curl_multi_perform () from /lib64/libcurl.so.4
#6  0x7f56726b75ca in resume_async_http_req (fd=9, msg=0x7f56739c0640 
, _param=0x7f56b3e354d0)
at rest_methods.c:380
#7  0x7f56737771ff in t_resume_async (fd=0x7f56b3e26840, 
param=0x7f567405c3e8) at async.c:125
#8  0x00509975 in handle_io (fm=0x7f56b3e26840, idx=1, event_type=1) at 
net/net_udp.c:267
#9  0x005082b3 in io_wait_loop_epoll (h=0x806e00 <_worker_io>, t=1, 
repeat=0) at net/../io_wait_loop.h:221
#10 0x00509c30 in udp_rcv_loop (si=0x7f56b3dd6808) at net/net_udp.c:308
#11 0x0050a1c8 in udp_start_processes (chd_rank=0x7d30e8 
, startup_done=0x0) at net/net_udp.c:372
#12 0x00492304 in main_loop () at main.c:671
#13 0x00494d8f in main (argc=7, argv=0x7fff38a979f8) at main.c:1261

Regards,
Agalya


From: Ramachandran, Agalya (Contractor)
Sent: Thursday, October 27, 2016 4:24 PM
To: users@lists.opensips.org; 'Liviu Chircu' 

Subject: Support for https in async

Hi team,

Just a quick question.. Does async(), method supports https request?
When I try to use https, resume_async_http_req is called, but I never got the 
response and rather OpenSIPS crashed at libcurl.

Regards,
Agalya

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Re: [OpenSIPS-Users] Support for https in async

[Liviu Chircu]

Hi Agalya!


handle https queries


Regarding HTTPSqueries, could you try to build your rest_client with 
"libcurl-openssl-dev" instead of "libcurl-gnutls-dev"as a possible fix?


> why is curl_multi_wait is not used

We don't want to block OpenSIPS after issuing an async query, but rather 
use the core async interface in order to quickly resume processing the 
next SIP message.



What is the use of read_fds[] in rest_methods.c


Helps us find the needle in the haystack. If we find the needle (i.e. 
the new fd), we can throw it into the reactor and successfully exit. 
Otherwise, the connect operation is still going, and we need to wait a 
bit more.



Why select() is not used ?


We don't want to do select() in the module code. The core reactor 
already does that for us. The module's only job in the async pattern is 
to produce file descriptors and feed them to the core async engine.


Regards,

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

On 08.11.2016 19:10, Ramachandran, Agalya (Contractor) wrote:


Hi Liviu,

Thank you for response. I have few questions, can you please clarify 
me on these.


·Is there is a roadmap in future targeting async() to handle https 
queries?


·In start_async_http_request(), why is curl_multi_wait is not used to 
overcome 1024 fds limitation? and is there plan to use it in future?


·What is the use of read_fds[] in rest_methods.c and who uses it?

·Why select() is not used ?

Regards,

Agalya

*From:*Liviu Chircu [mailto:li...@opensips.org]
*Sent:* Monday, October 31, 2016 12:27 PM
*To:* Ramachandran, Agalya (Contractor) 
; OpenSIPS users mailling list 


*Subject:* Re: Support for https in async

Hi, Agalya!

We have not done any development in that direction, and I assume it 
won't work out of the box, as we need to instruct libcurl on where to 
find the CA certificate before we can expect it to establish the TLS 
connection. As an alternative, we could add the option of disabling 
host/peer verifications like here [1]


[1]: https://curl.haxx.se/libcurl/c/https.html 



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

On 31.10.2016 16:41, Ramachandran, Agalya (Contractor) wrote:

Hi team,

Have you tried with *https://url*with async and does that work
without issue ?

When I try, resume_async_http_req is called, but crash is
occurring at libcurl library. So helpless to debug further.

async(rest_put(*"https://url"; , *"$fU,$tU,$ci ",
"application/json", "$var(body)", "$var(ct)", "$var(rcode)"),resume);

My question is crash occring only in my scenario or  OpenSIPS
doesn’t support async as https?

Here is my dump just in case for reference.

/#0 0x7f567248da1e in Curl_raw_nequal () from /lib64/libcurl.so.4/

/#1 0x7f567245bd0f in Curl_checkheaders () from
/lib64/libcurl.so.4/

/#2 0x7f567245d1e5 in Curl_http () from /lib64/libcurl.so.4/

/#3 0x7f567246db4b in Curl_do () from /lib64/libcurl.so.4/

/#4 0x7f567247da1b in multi_runsingle () from /lib64/libcurl.so.4/

/#5 0x7f567247e121 in curl_multi_perform () from
/lib64/libcurl.so.4/

#6 0x7f56726b75ca in resume_async_http_req (fd=9,
msg=0x7f56739c0640 , _param=0x7f56b3e354d0)

at rest_methods.c:380

#7 0x7f56737771ff in t_resume_async (fd=0x7f56b3e26840,
param=0x7f567405c3e8) at async.c:125

#8 0x00509975 in handle_io (fm=0x7f56b3e26840, idx=1,
event_type=1) at net/net_udp.c:267

#9 0x005082b3 in io_wait_loop_epoll (h=0x806e00
<_worker_io>, t=1, repeat=0) at net/../io_wait_loop.h:221

#10 0x00509c30 in udp_rcv_loop (si=0x7f56b3dd6808) at
net/net_udp.c:308

#11 0x0050a1c8 in udp_start_processes (chd_rank=0x7d30e8
, startup_done=0x0) at net/net_udp.c:372

#12 0x00492304 in main_loop () at main.c:671

#13 0x00494d8f in main (argc=7, argv=0x7fff38a979f8) at
main.c:1261

Regards,

Agalya

*From:* Ramachandran, Agalya (Contractor)
*Sent:* Thursday, October 27, 2016 4:24 PM
*To:* users@lists.opensips.org ;
'Liviu Chircu'  
*Subject:* Support for https in async

Hi team,

Just a quick question.. Does async(), method supports https request?

When I try to use https, resume_async_http_req is called, but I
never got the response and rather OpenSIPS crashed at libcurl.

Regards,
Agalya



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Re: [OpenSIPS-Users] Rtpproxy and IPV4 IPV6 interworking

[Robert Dyck]

Thank you

Assuming rtpproxy was started with IPV4 as the first address and IPV6 as the 
second, then in the NAT scenario, are the II flags mandatory in offer/answer?

Slightly off topic, what sort of scenario would require the address parameter 
for offer/answer?

On November 8, 2016 09:57:30 AM Răzvan Crainea wrote:
> Hi, Robert!
> 
> See my answers inline.
> 
> Best regards,
> 
> Răzvan Crainea
> OpenSIPS Solutions
> www.opensips-solutions.com
> 
> On 11/08/2016 02:15 AM, Robert Dyck wrote:
> > I have some question regarding rtpproxy capabilities in relation to
> > IPV4-IPV6 interworking.
> > 
> > The articles I have read say that you need to assign an address from each
> > address family to rtpproxy. They go on to say that rtpproxy will then be
> > in
> > bridged mode. Others define bridge mode as assigning two interfaces to
> > rtpproxy.
> 
> As long as you have RTPProxy listening on two IPs, you have it set in
> bridge mode. It doesn't matther whether one of them is IPv6, or both are.
> 
> > If the IPV4 and IPV6 addresses are on the same interface, is the rtpproxy
> > indeed in bridged mode? Should one avoid the use of engage_rtpproxy?
> 
> Yes, as stated above, RTPProxy is in bridged mode and you should avoid
> using engage_rtpproxy(). That's because the function can't know/decide
> which interface is which and cannot map with the RTPProxy's one.
> 
> > Assuming that IPV4- IPV6 interworking is actually possible using opensips
> > and rtpproxy, does that mean that an instance of rtpproxy is not
> > available to enable NAT traversal - would NAT traversal require using
> > another instance of rtpproxy using a single IPV4 address?
> 
> No, you don't need an extra instance - a single instance will do both
> bridging and nat traversal.
> 
> > Furthermore is the multihome parameter relevant to IPV4-IPV6 interworking
> > if opensips only listens on one interface?
> 
> The multihome parameter is only relevant for OpenSIPS, it doesn't
> influence RTPProxy's behavior at all.
> 
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[OpenSIPS-Users] How to make OpenSIPS send SIP BYE, by configuration, , before dialog timout?

[Rodrigo Pimenta Carvalho]

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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