[OpenSIPS-Users] Problem with aaa_radius

[Alexander Mustafin]

Hello.

I’m trying to setup new server and just copy opensips.cfg from other server. 
This configuration tested and works successfully.
But now, I’ve silent problem.

OpenSIPS from yum-repo: 
version: opensips 1.10.0beta-tls (x86_64/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, DISABLE_NAGLE, USE_MCAST, 
SHM_MEM, 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.
@(#) $Id$
main.c compiled on 03:46:24 Aug  6 2013 with gcc 4.4.7

radiusclient-ng installed and config copied too.

When I insert radius_send_auth("radius_auth","radius_auth_resp");   
 in config - OpenSIPS can’t start and I have no errors in syslog.
Sets are defined in module configuration.

Syslog:
===
Feb 24 02:34:50 reg opensips: DBG:core:yyparse: loading module 
/usr/lib64/opensips/modules/aaa_radius.so
Feb 24 02:34:50 reg opensips: DBG:core:set_mod_param_regex: aaa_radius matches 
module aaa_radius
Feb 24 02:34:50 reg opensips: DBG:core:set_mod_param_regex: found 
 in module aaa_radius [/usr/lib64/opensips/modules/]
Feb 24 02:34:50 reg opensips: DBG:core:set_mod_param_regex: aaa_radius matches 
module aaa_radius
Feb 24 02:34:50 reg opensips: DBG:core:set_mod_param_regex: found  in 
module aaa_radius [/usr/lib64/opensips/modules/]
Feb 24 02:34:50 reg opensips: DBG:core:__search_avp_map: looking for [username] 
avp  - found -1
Feb 24 02:34:50 reg opensips: DBG:core:new_avp_alias: added alias username with 
id 1
Feb 24 02:34:50 reg opensips: DBG:core:__search_avp_map: looking for [username] 
avp  - found 1
Feb 24 02:34:50 reg opensips: DBG:core:__search_avp_map: looking for [rpid] avp 
 - found -1
Feb 24 02:34:50 reg opensips: DBG:core:new_avp_alias: added alias rpid with id 2
Feb 24 02:34:50 reg opensips: DBG:core:set_mod_param_regex: aaa_radius matches 
module aaa_radius
Feb 24 02:34:50 reg opensips: DBG:core:set_mod_param_regex: found  in 
module aaa_radius [/usr/lib64/opensips/modules/]
Feb 24 02:34:50 reg opensips: DBG:core:__search_avp_map: looking for 
[credit_time] avp  - found -1
Feb 24 02:34:50 reg opensips: DBG:core:new_avp_alias: added alias credit_time 
with id 3
Feb 24 02:34:50 reg opensips: DBG:core:__search_avp_map: looking for 
[return_code] avp  - found -1
Feb 24 02:34:50 reg opensips: DBG:core:new_avp_alias: added alias return_code 
with id 4



Best regards,
Alexander Mustafin
mustafin.aleksa...@gmail.com






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Re: [OpenSIPS-Users] Truncating ACC and CDRs

[David Crow]

Yes, that’s the plan long term.  For now, though I’m just wondering if deleting 
everything in the acc table will cause problems.   Or does anyone have any 
other suggestions on how to clean this up?

-David

From: users-boun...@lists.opensips.org 
[mailto:users-boun...@lists.opensips.org] On Behalf Of Adrian Georgescu
Sent: Friday, February 21, 2014 6:31 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Truncating ACC and CDRs

Any operation that writes to your MySQL database for any great length of time 
like trimming your table will block anything else accesing the same database. 
You should auto rotate tables and use non transactional table storage to be 
able to trim older tables without impacting the active ones.

--
Adrian

On 21 Feb 2014, at 16:49, David Crow 
mailto:david.c...@vc3.com>> wrote:
I’ve got an opensips proxy that I just use for PSTN routing, no billing or 
anything is done from this system.
I only use the cdrs to identify when provider a call went to or from.

Both the ACC and CDRs table have just been growing and I haven’t done anything 
with them and now the acc table has gotten to over 4.6 GB and it’s almost 
impossible to search through.  I have stopped generating the CDRs for now.  I 
would like to just purge these tables and start over but I wasn’t sure if I 
would break anything if I run a truncate on the acc table.  Does anything read 
from the ACC table other than the CDR generation script?  Would it be ok to run 
a TRUNCATE on this table?

Thanks,

David Crow | Senior Systems Architect
1301 Gervais Street, Suite 1800 | Columbia, SC 29201
(d) 803.978.2727 | (f) 803.733.5888
david.c...@vc3.com| www.VC3.com
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Re: [OpenSIPS-Users] Initializing SIP messages from routing

[Răzvan Crainea]

Hi!

What do you mean the function does not work properly? It doesn't send 
any command to RTPProxy? Can you trace the communication between 
OpenSIPS and RTPProxy?
Also, have you checked the RTPProxy logs for errors? I am not sure how 
you can detect this, but if I remember correctly, I had to patch 
RTPProxy to properly inject media in early stage. If you want me, I can 
provide that patch for you so you can give it a try.


Best regards,

Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 02/24/2014 08:03 AM, H Yavari wrote:

Hi,
I used Opensips with RTPproxy and now RTP goes from RTPproxy. But
rtpproxy_stream2uac() function not work properly. my cfg:

if (is_method("INVITE")) {
 rtpproxy_offer("ro");
  rtpproxy_stream2uas("/var/rtpproxy/prompts/test", "-1");
 }

There isn't any info in logs about this function. what is the problem?




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[OpenSIPS-Users] Adding Proxy-Authorization header

[Diego Barberio]

Hi all,

I have opensips registered to an IP-PBX using registrant module and I want
to make an outbound call to that PBX through the proxy.

I'm sending and INVITE from my application to the proxy with a From that is
actually registered by the proxy, however OpenSIPs is not adding the
Proxy-Authorization header so the INVITE is rejected with a 401
Unauthorized and that response is forwarded to my application.

I just want opensips to add the Proxy-Authorization header so the call is
not rejected by the IP-PBX. Is it possible to achieve this?

Thanks
Diego
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Re: [OpenSIPS-Users] Adding Proxy-Authorization header

[Stefano Pisani]

You can use module UAC_AUTH

Il 24/02/2014 16.18, Diego Barberio ha scritto:

Hi all,

I have opensips registered to an IP-PBX using registrant module and I 
want to make an outbound call to that PBX through the proxy.


I'm sending and INVITE from my application to the proxy with a From 
that is actually registered by the proxy, however OpenSIPs is not 
adding the Proxy-Authorization header so the INVITE is rejected with a 
401 Unauthorized and that response is forwarded to my application.


I just want opensips to add the Proxy-Authorization header so the call 
is not rejected by the IP-PBX. Is it possible to achieve this?


Thanks
Diego


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Re: [OpenSIPS-Users] Adding Proxy-Authorization header

[Vlad Paiu]

Hello,

The registrant module is to be used only for generating REGISTER 
requests ( with auth included ).
For proxied calls, you need to use the uac and uac_auth modules ( [1] ) 
for adding the auth headers - call uac_auth() ( [2] ) function within 
failure route when receiving a challenge.


[1] http://www.opensips.org/html/docs/modules/1.11.x/uac_auth.html
[2] http://www.opensips.org/html/docs/modules/1.11.x/uac.html#id250288

Best Regards

Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com

On 24.02.2014 17:33, Stefano Pisani wrote:

You can use module UAC_AUTH

Il 24/02/2014 16.18, Diego Barberio ha scritto:

Hi all,

I have opensips registered to an IP-PBX using registrant module and I 
want to make an outbound call to that PBX through the proxy.


I'm sending and INVITE from my application to the proxy with a From 
that is actually registered by the proxy, however OpenSIPs is not 
adding the Proxy-Authorization header so the INVITE is rejected with 
a 401 Unauthorized and that response is forwarded to my application.


I just want opensips to add the Proxy-Authorization header so the 
call is not rejected by the IP-PBX. Is it possible to achieve this?


Thanks
Diego


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[OpenSIPS-Users] opensips b2b $ci set

[M.Khaled W Chehab]

HI

I am using b2buac module with topology hidding  is there a way to set  the
$ci  instead of let opensips generate it  for ex:  b2b_reply
(B2B.361.6489307)  i want it to be b2b_reply (B2B.361.opensips)

 

 

route[b2b_request] {

  xlog("b2b_request ($ci)\n");

}

 

 

route[b2b_reply] {

  xlog("b2b_reply ($ci)\n");

}

 

 

 

route[1] {

# for INVITEs enable some additional helper routes

 

if (is_method("INVITE") && !(src_ip == "

 

  b2b_init_request("top hiding");

 

Regards

 

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[OpenSIPS-Users] Opensips1.7 with MediaProxy

[Лытаев Антон Викторович]

Hi. Please help.
We have:
1.One server consists of: CenOS6.5 + Opensips1.7 + MediaProxy2.5
2.One MGW: Cisco AS5350
3.UserID=telephone number and registration on OpenSips through MySQL
4.Call to PSTN pass through MGW with prefix :
route[4] {  prefix("");
  rewritehostport("192.168.0.3:5060 ");
  if (!t_relay()) { sl_reply_error(); };
exit;}

Now, such a scheme works:

(UAC   )>sip->Opensips 1.7--->SIP--->MGW Cisco
192.168.0.65   192.168.0.2 192.168.0.3
RTP---
>MGW Cisco>PSTN

In this topology visible

It's not safe, it's necessary to build a new wiring diagram:
(UAC  )--->sip,RTP>(Opensips--->rtp,SIP-->)->MGW 
Cisco--->PSTN

85.85.85.85(85.85.85.2 192.168.0.2) 192.168.0.3

questions:
1. to hide the network topology from the users (can be used dialog 
module, function: topology_hiding?)
2. hide RTP traffic to MGW for Opensips-server (can be used MediaProxy 
or rtpproxy)?

Please, give examples opensips.cfg-file ?
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