Re: [OpenSIPS-Users] Branch Limit

[Mike Tesliuk]

Thanks, 


I will do some more tests to understand why was not complete, but i
solve the problem over failure_route.


Thanks Răzvan


Em 26/09/17 13:26, Răzvan Crainea escreveu:
> Hi, Mike!
>
> You can append up to 12 branches.
>
> Best regards,
> Răzvan Crainea
> OpenSIPS Developer
> www.opensips-solutions.com
> On 09/26/2017 06:19 PM, Mike Tesliuk wrote:
>> Hello,
>>
>> Somebody can confirm if i can append just 3 branches to a call ?
>>
>> I have a situation that i have to do a pickup search on multiple
>> asterisk's , i was trying to use appen_branch for that, but just 3 of 6
>> of my pbx are being reached .
>>
>> Thank you.
>>
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

-- 


​Atenciosamente,
WSU TECNOLOGIA
Mike Tesliuk
Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR  
12387 SW 125th ter, Miami, Florida 33186 - USA
tel +55 (41) 3941.0650   +1 (786) 719.6253
*website <http://www.wsu.com.br/>  |  mapa
<https://www.google.com.br/maps/place/WSU+Tecnologia/@-25.4354389,-49.2779048,17z/data=%213m1%214b1%214m2%213m1%211s0x94dce473a24cf705:0x369fdf05247b568b?hl=pt-BR>
  |  email
<mailto:cont...@wsu.com.br>    *
.



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[OpenSIPS-Users] Branch Limit

[Mike Tesliuk]

Hello,

Somebody can confirm if i can append just 3 branches to a call ?

I have a situation that i have to do a pickup search on multiple
asterisk's , i was trying to use appen_branch for that, but just 3 of 6
of my pbx are being reached .

Thank you.



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Re: [OpenSIPS-Users] pv_proxy_authorize cache

[Mike Tesliuk]

Hello Bogdan,  the $avp(usuario) is populated with the $fU as you think,
the password is using the calculate parameter , but my question was if
the pv_proxy_authorize is supposed to work, because on the example was
used the www_ , and on my tests  do not work, but i will double check my
configuration an try again.


thank you very much


Em 25/07/17 09:45, Bogdan-Andrei Iancu escreveu:
> Hi Mike,
>
> depending on your SIP flow, you can use either www_ (if a REGISTER) or
> proxy_ (if a non-REGISTER) functions.
>
> In your script snip, you must populate both auth username and password
> before the calling the auth function. I do not see the $avp(usuario)
> set (probably with $fU ??) . Also, if the password is plain/text, be
> use you properly set the calculate_ha1 parameter.
>
> Best regards,
> Bogdan-Andrei Iancu
>   OpenSIPS Founder and Developer
>   http://www.opensips-solutions.com
>
> OpenSIPS Bootcamp 2017, Houston, US
>   http://opensips.org/training/OpenSIPS_Bootcamp_2017.html
> On 07/24/2017 11:21 PM, Mike Tesliuk wrote:
>>
>> Hello there,
>>
>>
>> Im trying to implement a proxy_authorize using cache without success.
>>
>>
>> is that possible to perform the cache using proxy_authorize ? on the
>> example [1] i see the www_challenge() no proxy_challenge, is that
>> correct ?
>>
>>
>> on my test im doing this (below):
>>
>>
>> modparam("auth","username_spec", "$avp(usuario)")
>> modparam("auth","password_spec", "$avp(senha)")
>> modparam("auth_db", "load_credentials", "$avp(senha)=password")
>>
>>
>> $avp(usuario) = $fU;
>>
>> if(cache_fetch("redis","passwd_$fU",$avp(senha))) {
>>  if(!pv_proxy_authorize("")){
>>  proxy_challenge("","0");
>>  exit;
>>  }
>> }else{
>>  if(!proxy_authorize("")){
>>  proxy_challenge("","0");
>>  exit;
>>  }
>>
>>  
>>  cache_store("redis","passwd_$fU","$avp(senha)",3600);
>>
>> }
>>
>> But with this rule i do not get the user authenticated.
>>
>> what im doing wrong ? :)
>>
>>
>> Thanks in advice 
>>
>>
>>
>>
>> [1] - https://www.opensips.org/Documentation/Tutorials-MemoryCaching#toc3
>>
>> -- 
>>
>>
>> ​Atenciosamente,
>> WSU TECNOLOGIA
>> Mike Tesliuk
>> Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR  
>> 12387 SW 125th ter, Miami, Florida 33186 - USA
>> tel +55 (41) 3941.0650   +1 (786) 719.6253
>> *website <http://www.wsu.com.br/>  |  mapa
>> <https://www.google.com.br/maps/place/WSU+Tecnologia/@-25.4354389,-49.2779048,17z/data=%213m1%214b1%214m2%213m1%211s0x94dce473a24cf705:0x369fdf05247b568b?hl=pt-BR>
>>   |  email
>> <mailto:cont...@wsu.com.br>*
>> .
>>
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

-- 


​Atenciosamente,
WSU TECNOLOGIA
Mike Tesliuk
Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR  
12387 SW 125th ter, Miami, Florida 33186 - USA
tel +55 (41) 3941.0650   +1 (786) 719.6253
*website <http://www.wsu.com.br/>  |  mapa
<https://www.google.com.br/maps/place/WSU+Tecnologia/@-25.4354389,-49.2779048,17z/data=%213m1%214b1%214m2%213m1%211s0x94dce473a24cf705:0x369fdf05247b568b?hl=pt-BR>
  |  email
<mailto:cont...@wsu.com.br>*
.



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Re: [OpenSIPS-Users] Registered trunks

[Mike Tesliuk]

Hello Pat,


I think that you can ask them to set the From Name as the callerid so
you can use transformation to take de information [1]


Example:

xlog("FROM NAME: $(hdr(From){nameaddr.name})");
xlog("FROM USER: $fU");

   

Result (on log):


l 24 22:06:51 opensipsHomolog2 /usr/local/sbin/opensips[7960]: FROM
NAME: "1016"
Jul 24 22:06:51 opensipsHomolog2 /usr/local/sbin/opensips[7960]: FROM
USER: 10160393



After the authentication you can use the uac_replace_from[2] and change
the callerid that you send you carriers.



[1] - https://www.opensips.org/Documentation/Script-Tran-2-2
[2] - http://www.opensips.org/html/docs/modules/devel/uac.html#idp5265536


Em 24/07/17 21:32, Pat Burke escreveu:
> Hello,
>
> As a SIP Provider, we implementing the ability to provide SIP trunks
> to customers with a PBX or Dialer that require Registration.  With
> this in mind,
> the customer wants to be able to set the CallerID on at least on the
> basis of the devices connected tho them, but potentially on a per call
> basis.
>
> For the challenge-response to the non-Register methods, we have
> implemented the script as follows (seems to be a very standard way). 
> My question is
> for the case of the CallerID not being the same as the
> username/authorization name, how do we do this?  Because the "FROM"
> user is different from the
> authorized user, the db_check_from fails.  I don't believe all phone
> systems support P-Asserted-ID, so we can't really go that route.  So
> can we just remove
> the "db_check_from"?   What risk does that expose us to?
>
> if ( !(is_method("REGISTER")) ) {
>   if (is_from_local("$var(reg_domain_attr)")) { # from Registered device
>   $avp(callee_number_type) := "Registered";
>
>   # authenticate if from local subscriber
>   # authenticate all initial non-REGISTER request that pretend to be
>   # generated by local subscriber (domain from FROM URI is local)
>   if (!proxy_authorize("", "subscriber")) {
>  proxy_challenge("", "0");
>  exit;
>   }
>
>   if (!db_check_from()) {
>  sl_send_reply("403","Forbidden auth ID");
>  exit;
>   }
>
>   consume_credentials();
>   # caller authenticated
>}
> }
> Regards,
> *Pat Burke*
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

-- 


​Atenciosamente,
WSU TECNOLOGIA
Mike Tesliuk
Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR  
12387 SW 125th ter, Miami, Florida 33186 - USA
tel +55 (41) 3941.0650   +1 (786) 719.6253
*website <http://www.wsu.com.br/>  |  mapa
<https://www.google.com.br/maps/place/WSU+Tecnologia/@-25.4354389,-49.2779048,17z/data=%213m1%214b1%214m2%213m1%211s0x94dce473a24cf705:0x369fdf05247b568b?hl=pt-BR>
  |  email
<mailto:cont...@wsu.com.br>*
.



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[OpenSIPS-Users] pv_proxy_authorize cache

[Mike Tesliuk]

Hello there,


Im trying to implement a proxy_authorize using cache without success.


is that possible to perform the cache using proxy_authorize ? on the
example [1] i see the www_challenge() no proxy_challenge, is that correct ?


on my test im doing this (below):


modparam("auth","username_spec", "$avp(usuario)")
modparam("auth","password_spec", "$avp(senha)")
modparam("auth_db", "load_credentials", "$avp(senha)=password")


$avp(usuario) = $fU;

if(cache_fetch("redis","passwd_$fU",$avp(senha))) {
if(!pv_proxy_authorize("")){
proxy_challenge("","0");
exit;
}
}else{
if(!proxy_authorize("")){
proxy_challenge("","0");
exit;
}


cache_store("redis","passwd_$fU","$avp(senha)",3600);

}

But with this rule i do not get the user authenticated.

what im doing wrong ? :)


Thanks in advice 




[1] - https://www.opensips.org/Documentation/Tutorials-MemoryCaching#toc3

-- 


​Atenciosamente,
WSU TECNOLOGIA
Mike Tesliuk
Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR  
12387 SW 125th ter, Miami, Florida 33186 - USA
tel +55 (41) 3941.0650   +1 (786) 719.6253
*website <http://www.wsu.com.br/>  |  mapa
<https://www.google.com.br/maps/place/WSU+Tecnologia/@-25.4354389,-49.2779048,17z/data=%213m1%214b1%214m2%213m1%211s0x94dce473a24cf705:0x369fdf05247b568b?hl=pt-BR>
  |  email
<mailto:cont...@wsu.com.br>*
.



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Re: [OpenSIPS-Users] Error on WSS call with RTPEngine

[Mike Tesliuk]

My bad Razvan,

The sceneario was different, but is the same now


http://sip.wsu.com.br/pub/test2.txt


Em 24/07/17 11:32, Răzvan Crainea escreveu:
> Hi, Mike!
>
> The debug log you have posted does not contain any CRITICAL or ERROR
> message in it. Was it done with a different scenario? If so, can you
> post the debug logs from the scenario that generates the CRITICAL message?
>
> Best regards,
> Răzvan Crainea
> OpenSIPS Solutions
> www.opensips-solutions.com
> On 07/24/2017 03:44 PM, Mike Tesliuk wrote:
>>
>> Hello Razvan ,
>>
>>
>> i have a debug log here:
>>
>>
>> http://sip.wsu.com.br/pub/test1.txt
>>
>>
>> This was a try of call fro 10150393 to 1017 , the call need to go
>> to an asterisk, and reach the extension on another opensips, this
>> work well with two simple softphone, but got the error when i try to
>> use the rtpengine
>>
>>
>> the same message as i send on another email happen when i try from
>> ipv4 to ipv6 using rtpproxy
>>
>>
>>
>>
>> Em 24/07/17 04:47, Răzvan Crainea escreveu:
>>> Hi, Mike!
>>>
>>> Can you send us the debugging log for this error?
>>>
>>> Best regards,
>>> Răzvan Crainea
>>> OpenSIPS Solutions
>>> www.opensips-solutions.com
>>> On 07/23/2017 06:49 PM, Mike Tesliuk wrote:
>>>>
>>>> Hello Guys,
>>>>
>>>>
>>>> On my tests with WSS call, im trying to go trough an asterisk and
>>>> cameback to another extension, when this happen the opensips crash
>>>> and show on log the message below
>>>>
>>>>
>>>> Jul 23 11:44:44 opensipsHomolog /usr/local/sbin/opensips[11065]:
>>>> CRITICAL:core:build_res_buf_from_sip_res: #012>>> len mistmatch :
>>>> calculated 1271, written 1237#012#012It seems you have hit a
>>>> programming bug.#012Please help us make OpenSIPS better by
>>>> reporting it at https://github.com/OpenSIPS/opensips/issues
>>>>
>>>>
>>>> This happen on the answer of the call
>>>>
>>>>
>>>> To you guys have any tip about this question ?
>>>>
>>>>
>>>> Thank you.
>>>>
>>>> -- 
>>>>
>>>>
>>>> ​Atenciosamente,
>>>> WSU TECNOLOGIA
>>>> Mike Tesliuk
>>>> Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR  
>>>> 12387 SW 125th ter, Miami, Florida 33186 - USA
>>>> tel +55 (41) 3941.0650   +1 (786) 719.6253
>>>> *website <http://www.wsu.com.br/>  |  mapa
>>>> <https://www.google.com.br/maps/place/WSU+Tecnologia/@-25.4354389,-49.2779048,17z/data=%213m1%214b1%214m2%213m1%211s0x94dce473a24cf705:0x369fdf05247b568b?hl=pt-BR>
>>>>   |  email
>>>> <mailto:cont...@wsu.com.br>*
>>>> .
>>>>
>>>>
>>>>
>>>> ___
>>>> Users mailing list
>>>> Users@lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>> -- 
>>
>>
>> ​Atenciosamente,
>> WSU TECNOLOGIA
>> Mike Tesliuk
>> Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR  
>> 12387 SW 125th ter, Miami, Florida 33186 - USA
>> tel +55 (41) 3941.0650   +1 (786) 719.6253
>> *website <http://www.wsu.com.br/>  |  mapa
>> <https://www.google.com.br/maps/place/WSU+Tecnologia/@-25.4354389,-49.2779048,17z/data=%213m1%214b1%214m2%213m1%211s0x94dce473a24cf705:0x369fdf05247b568b?hl=pt-BR>
>>   |  email
>> <mailto:cont...@wsu.com.br>*
>> .
>>
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

-- 


​Atenciosamente,
WSU TECNOLOGIA
Mike Tesliuk
Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR  
12387 SW 125th ter, Miami, Florida 33186 - USA
tel +55 (41) 3941.0650   +1 (786) 719.6253
*website <http://www.wsu.com.br/>  |  mapa
<https://www.google.com.br/maps/place/WSU+Tecnologia/@-25.4354389,-49.2779048,17z/data=%213m1%214b1%214m2%213m1%211s0x94dce473a24cf705:0x369fdf05247b568b?hl=pt-BR>
  |  email
<mailto:cont...@wsu.com.br>*
.



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Re: [OpenSIPS-Users] Error on WSS call with RTPEngine

[Mike Tesliuk]

Hello Razvan ,


i have a debug log here:


http://sip.wsu.com.br/pub/test1.txt


This was a try of call fro 10150393 to 1017 , the call need to go to
an asterisk, and reach the extension on another opensips, this work well
with two simple softphone, but got the error when i try to use the rtpengine


the same message as i send on another email happen when i try from ipv4
to ipv6 using rtpproxy




Em 24/07/17 04:47, Răzvan Crainea escreveu:
> Hi, Mike!
>
> Can you send us the debugging log for this error?
>
> Best regards,
> Răzvan Crainea
> OpenSIPS Solutions
> www.opensips-solutions.com
> On 07/23/2017 06:49 PM, Mike Tesliuk wrote:
>>
>> Hello Guys,
>>
>>
>> On my tests with WSS call, im trying to go trough an asterisk and
>> cameback to another extension, when this happen the opensips crash
>> and show on log the message below
>>
>>
>> Jul 23 11:44:44 opensipsHomolog /usr/local/sbin/opensips[11065]:
>> CRITICAL:core:build_res_buf_from_sip_res: #012>>> len mistmatch :
>> calculated 1271, written 1237#012#012It seems you have hit a
>> programming bug.#012Please help us make OpenSIPS better by reporting
>> it at https://github.com/OpenSIPS/opensips/issues
>>
>>
>> This happen on the answer of the call
>>
>>
>> To you guys have any tip about this question ?
>>
>>
>> Thank you.
>>
>> -- 
>>
>>
>> ​Atenciosamente,
>> WSU TECNOLOGIA
>> Mike Tesliuk
>> Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR  
>> 12387 SW 125th ter, Miami, Florida 33186 - USA
>> tel +55 (41) 3941.0650   +1 (786) 719.6253
>> *website <http://www.wsu.com.br/>  |  mapa
>> <https://www.google.com.br/maps/place/WSU+Tecnologia/@-25.4354389,-49.2779048,17z/data=%213m1%214b1%214m2%213m1%211s0x94dce473a24cf705:0x369fdf05247b568b?hl=pt-BR>
>>   |  email
>> <mailto:cont...@wsu.com.br>*
>> .
>>
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

-- 


​Atenciosamente,
WSU TECNOLOGIA
Mike Tesliuk
Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR  
12387 SW 125th ter, Miami, Florida 33186 - USA
tel +55 (41) 3941.0650   +1 (786) 719.6253
*website <http://www.wsu.com.br/>  |  mapa
<https://www.google.com.br/maps/place/WSU+Tecnologia/@-25.4354389,-49.2779048,17z/data=%213m1%214b1%214m2%213m1%211s0x94dce473a24cf705:0x369fdf05247b568b?hl=pt-BR>
  |  email
<mailto:cont...@wsu.com.br>*
.



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Re: [OpenSIPS-Users] Error on WSS call with RTPEngine

[Mike Tesliuk]

The same error when i try a ipv6 -> ipv4 using rtpproxy


Em 23/07/17 12:49, Mike Tesliuk escreveu:
>
> Hello Guys,
>
>
> On my tests with WSS call, im trying to go trough an asterisk and
> cameback to another extension, when this happen the opensips crash and
> show on log the message below
>
>
> Jul 23 11:44:44 opensipsHomolog /usr/local/sbin/opensips[11065]:
> CRITICAL:core:build_res_buf_from_sip_res: #012>>> len mistmatch :
> calculated 1271, written 1237#012#012It seems you have hit a
> programming bug.#012Please help us make OpenSIPS better by reporting
> it at https://github.com/OpenSIPS/opensips/issues
>
>
> This happen on the answer of the call
>
>
> To you guys have any tip about this question ?
>
>
> Thank you.
>
> -- 
>
>
> ​Atenciosamente,
> WSU TECNOLOGIA
> Mike Tesliuk
> Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR  
> 12387 SW 125th ter, Miami, Florida 33186 - USA
> tel +55 (41) 3941.0650   +1 (786) 719.6253
> *website <http://www.wsu.com.br/>  |  mapa
> <https://www.google.com.br/maps/place/WSU+Tecnologia/@-25.4354389,-49.2779048,17z/data=%213m1%214b1%214m2%213m1%211s0x94dce473a24cf705:0x369fdf05247b568b?hl=pt-BR>
>   |  email
> <mailto:cont...@wsu.com.br>*
> .
>

-- 


​Atenciosamente,
WSU TECNOLOGIA
Mike Tesliuk
Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR  
12387 SW 125th ter, Miami, Florida 33186 - USA
tel +55 (41) 3941.0650   +1 (786) 719.6253
*website <http://www.wsu.com.br/>  |  mapa
<https://www.google.com.br/maps/place/WSU+Tecnologia/@-25.4354389,-49.2779048,17z/data=%213m1%214b1%214m2%213m1%211s0x94dce473a24cf705:0x369fdf05247b568b?hl=pt-BR>
  |  email
<mailto:cont...@wsu.com.br>*
.



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[OpenSIPS-Users] Error on WSS call with RTPEngine

[Mike Tesliuk]

Hello Guys,


On my tests with WSS call, im trying to go trough an asterisk and
cameback to another extension, when this happen the opensips crash and
show on log the message below


Jul 23 11:44:44 opensipsHomolog /usr/local/sbin/opensips[11065]:
CRITICAL:core:build_res_buf_from_sip_res: #012>>> len mistmatch :
calculated 1271, written 1237#012#012It seems you have hit a programming
bug.#012Please help us make OpenSIPS better by reporting it at
https://github.com/OpenSIPS/opensips/issues


This happen on the answer of the call


To you guys have any tip about this question ?


Thank you.

-- 


​Atenciosamente,
WSU TECNOLOGIA
Mike Tesliuk
Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR  
12387 SW 125th ter, Miami, Florida 33186 - USA
tel +55 (41) 3941.0650   +1 (786) 719.6253
*website <http://www.wsu.com.br/>  |  mapa
<https://www.google.com.br/maps/place/WSU+Tecnologia/@-25.4354389,-49.2779048,17z/data=%213m1%214b1%214m2%213m1%211s0x94dce473a24cf705:0x369fdf05247b568b?hl=pt-BR>
  |  email
<mailto:cont...@wsu.com.br>*
.



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Re: [OpenSIPS-Users] Permission denied bo bind port 443 or 80

[Mike Tesliuk]

As i told i already have done that, the question is because one of the
documentation have used the port 443


http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2



but that is not a problems i was just trying to understood


Em 23/07/17 12:24, David Villasmil escreveu:
> There's some permission restrictions that won't allow a non-root user
> to bind to those ports. Have a look at the OS documentation to figure
> out how to allow that... don't know what OS you're on
> On Sun, Jul 23, 2017 at 4:18 PM Mike Tesliuk  <mailto:m...@wsu.com.br>> wrote:
>
> Hello,
>
> Im creating an enviroment with TLS and WSS and i got permission
> denied when trying to start the wss and ws using port 80 or 443
>
>
> Jul 23 10:04:40 opensipsHomolog /usr/local/sbin/opensips[3494]:
> ERROR:core:tcp_init_listener: bind(c, 0x7efca6dc1e5c, 16) on
> 168.194.68.29:443 <http://168.194.68.29:443> : Permission denied
> Jul 23 10:04:40 opensipsHomolog /usr/local/sbin/opensips[3494]:
> ERROR:core:trans_init_all_listeners: failed to init listener
> [168.194.68.29], proto wss
>
>
> This occur why im running opensips as a user (opensips) and not as
> root, there is a setcap option that can allow this to happen, but,
> i think that this is some kind of mistake on my configuration, im
> right ? as we got daemons like nginx or apache that run as user
> but can have use of those ports , how can i do the same on opensips ?
>
>
> Thank you
>
>
> -- 
>
>
> ​Atenciosamente,
> WSU TECNOLOGIA
> Mike Tesliuk
> Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR  
> 12387 SW 125th ter, Miami, Florida 33186 - USA
> tel +55 (41) 3941.0650   +1 (786) 719.6253
> *website <http://www.wsu.com.br/>  |  mapa
> 
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​Atenciosamente,
WSU TECNOLOGIA
Mike Tesliuk
Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR  
12387 SW 125th ter, Miami, Florida 33186 - USA
tel +55 (41) 3941.0650   +1 (786) 719.6253
*website <http://www.wsu.com.br/>  |  mapa
<https://www.google.com.br/maps/place/WSU+Tecnologia/@-25.4354389,-49.2779048,17z/data=%213m1%214b1%214m2%213m1%211s0x94dce473a24cf705:0x369fdf05247b568b?hl=pt-BR>
  |  email
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[OpenSIPS-Users] Permission denied bo bind port 443 or 80

[Mike Tesliuk]

Hello,

Im creating an enviroment with TLS and WSS and i got permission denied
when trying to start the wss and ws using port 80 or 443


Jul 23 10:04:40 opensipsHomolog /usr/local/sbin/opensips[3494]:
ERROR:core:tcp_init_listener: bind(c, 0x7efca6dc1e5c, 16) on
168.194.68.29:443 : Permission denied
Jul 23 10:04:40 opensipsHomolog /usr/local/sbin/opensips[3494]:
ERROR:core:trans_init_all_listeners: failed to init listener
[168.194.68.29], proto wss


This occur why im running opensips as a user (opensips) and not as root,
there is a setcap option that can allow this to happen, but, i think
that this is some kind of mistake on my configuration, im right ? as we
got daemons like nginx or apache that run as user but can have use of
those ports , how can i do the same on opensips ?


Thank you


-- 


​Atenciosamente,
WSU TECNOLOGIA
Mike Tesliuk
Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR  
12387 SW 125th ter, Miami, Florida 33186 - USA
tel +55 (41) 3941.0650   +1 (786) 719.6253
*website <http://www.wsu.com.br/>  |  mapa
<https://www.google.com.br/maps/place/WSU+Tecnologia/@-25.4354389,-49.2779048,17z/data=%213m1%214b1%214m2%213m1%211s0x94dce473a24cf705:0x369fdf05247b568b?hl=pt-BR>
  |  email
<mailto:cont...@wsu.com.br>*
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[OpenSIPS-Users] LoadBalancer and Clusterer

[Mike Tesliuk]

Hello,


On the past, i had implemented the dialog with cachedb and load_balancer
using a nosql, using the resource with the /s , as the load_balancer
have the parameter receive the replication, how i use that ?


without the /s when i create a call i do not se the resource being used
on node 2 , is that supposed to happen ?


Thank you.


PS: testing the 2.3 version


-- 


​Atenciosamente,
WSU TECNOLOGIA
Mike Tesliuk
Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR  
12387 SW 125th ter, Miami, Florida 33186 - USA
tel +55 (41) 3941.0650   +1 (786) 719.6253
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  |  email
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Re: [OpenSIPS-Users] Call continuity

[Mike Tesliuk]

This is the kind of structure where you need a BGP session with your
carriers, when you have an ASN and your own IP Block you can have your
communication flowing between any internet links you want, as the ip
address will flow between all of them, this is the right way.


if you have two simples internet connections there is no way to do that


Em 20/07/17 17:11, Alex Balashov escreveu:
> On Thu, Jul 20, 2017 at 08:07:28PM +, Abdirahman A. Osman wrote:
>
>> Is it possible to keep a live call continue , if the internet
>> connection fails and route it through another internet connection ?
>> Does SIP protocol support this kind of Call continuity?
> Generally speaking, no, though the answer will vary with the modalities
> of the failover mechanism.
>
> But in general, failing anything over to another Internet connection
> means changing the address of one of the endpoints involved. All
> session-based Internet applications, whether using a
> connection-orientated transport or not, presume that the IP and port
> endpoints on both ends stay the same. 
>
> So, if you suddenly start sending media from another place and expecting
> to receive it there likewise, that will not be considered to be part of
> the same phone call.
>
> -- Alex
>

-- 


​Atenciosamente,
WSU TECNOLOGIA
Mike Tesliuk
Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR  
12387 SW 125th ter, Miami, Florida 33186 - USA
tel +55 (41) 3941.0650   +1 (786) 719.6253
*website <http://www.wsu.com.br/>  |  mapa
<https://www.google.com.br/maps/place/WSU+Tecnologia/@-25.4354389,-49.2779048,17z/data=%213m1%214b1%214m2%213m1%211s0x94dce473a24cf705:0x369fdf05247b568b?hl=pt-BR>
  |  email
<mailto:cont...@wsu.com.br>*
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[OpenSIPS-Users] error on script generation

[Mike Tesliuk]

Hello there,


im compiling the opensips 2.3 and the generated script with tls generate
the lines below:

modparam("proto_tls","verify_cert", "1")
modparam("proto_tls","require_cert", "0")
modparam("proto_tls","tls_method", "TLSv1")

modparam("proto_tls","certificate",
"/usr/local/etc/opensips/tls/user/user-cert.pem")
modparam("proto_tls","private_key",
"/usr/local/etc/opensips/tls/user/user-privkey.pem")
modparam("proto_tls","ca_list",
"/usr/local/etc/opensips/tls/user/user-calist.pem")


Those parameters are from tls_mgm not from proto_tls right ? on module
documentation are on tls_mgm section


-- 


​Atenciosamente,
WSU TECNOLOGIA
Mike Tesliuk
Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR  
12387 SW 125th ter, Miami, Florida 33186 - USA
tel +55 (41) 3941.0650   +1 (786) 719.6253
*website <http://www.wsu.com.br/>  |  mapa
<https://www.google.com.br/maps/place/WSU+Tecnologia/@-25.4354389,-49.2779048,17z/data=%213m1%214b1%214m2%213m1%211s0x94dce473a24cf705:0x369fdf05247b568b?hl=pt-BR>
  |  email
<mailto:cont...@wsu.com.br>*
.



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Re: [OpenSIPS-Users] Error in compile of tls_mgm

[Mike Tesliuk]

i got a Hunk


[2017-07-22 08:07:33] root@opensipsHomolog /usr/src/opensips-2.2 # 
patch -p1 < /root/port-tls-1.1.0.patch
patching file modules/tls_mgm/tls.h
patching file modules/tls_mgm/tls_conn_ops.h
patching file modules/tls_mgm/tls_conn_server.h
patching file modules/tls_mgm/tls_mgm.c
Hunk #3 succeeded at 1154 (offset 81 lines).
Hunk #4 succeeded at 1178 (offset 81 lines).
Hunk #5 succeeded at 1229 (offset 81 lines).
Hunk #6 succeeded at 1239 (offset 81 lines).
Hunk #7 succeeded at 1388 (offset 80 lines).
Hunk #8 succeeded at 1406 (offset 80 lines).
Hunk #9 succeeded at 1436 with fuzz 1 (offset 80 lines).
patching file modules/identity/identity.c


i will  try to apply mannually and let you know


Em 21/07/17 13:49, Răzvan Crainea escreveu:
> Hi, Mike!
>
> You can apply this[1] patch to make opensips 2.2 compatible with ssl
> 1.1.0. However, this has not yet been fully tested, that's why it's
> not present in the current version.
> Please give it a try and let me know how it goes.
>
> [1]
> https://sources.debian.net/src/opensips/2.2.2-3/debian/patches/port-tls-1.1.0.patch/
>
> Best regards,
> Răzvan Crainea
> OpenSIPS Solutions
> www.opensips-solutions.com
> On 07/21/2017 07:16 PM, Mike Tesliuk wrote:
>>
>> I got more errors on this case.
>>
>>
>> i will try the 2.3 version, thank  you.
>>
>>
>> Em 21/07/17 11:54, Răzvan Crainea escreveu:
>>> CC_EXTRA_OPTS=-DOPENSSL_NO_KRB5 make modules modules=modules/tls_mgm
>>
>> -- 
>>
>>
>> ​Atenciosamente,
>> WSU TECNOLOGIA
>> Mike Tesliuk
>> Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR  
>> 12387 SW 125th ter, Miami, Florida 33186 - USA
>> tel +55 (41) 3941.0650   +1 (786) 719.6253
>> *website <http://www.wsu.com.br/>  |  mapa
>> <https://www.google.com.br/maps/place/WSU+Tecnologia/@-25.4354389,-49.2779048,17z/data=%213m1%214b1%214m2%213m1%211s0x94dce473a24cf705:0x369fdf05247b568b?hl=pt-BR>
>>   |  email
>> <mailto:cont...@wsu.com.br>*
>> .
>>
>>
>>
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>
>
>
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-- 


​Atenciosamente,
WSU TECNOLOGIA
Mike Tesliuk
Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR  
12387 SW 125th ter, Miami, Florida 33186 - USA
tel +55 (41) 3941.0650   +1 (786) 719.6253
*website <http://www.wsu.com.br/>  |  mapa
<https://www.google.com.br/maps/place/WSU+Tecnologia/@-25.4354389,-49.2779048,17z/data=%213m1%214b1%214m2%213m1%211s0x94dce473a24cf705:0x369fdf05247b568b?hl=pt-BR>
  |  email
<mailto:cont...@wsu.com.br>*
.



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Re: [OpenSIPS-Users] Error in compile of tls_mgm

[Mike Tesliuk]

I got more errors on this case.


i will try the 2.3 version, thank  you.


Em 21/07/17 11:54, Răzvan Crainea escreveu:
> CC_EXTRA_OPTS=-DOPENSSL_NO_KRB5 make modules modules=modules/tls_mgm

-- 


​Atenciosamente,
WSU TECNOLOGIA
Mike Tesliuk
Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR  
12387 SW 125th ter, Miami, Florida 33186 - USA
tel +55 (41) 3941.0650   +1 (786) 719.6253
*website <http://www.wsu.com.br/>  |  mapa
<https://www.google.com.br/maps/place/WSU+Tecnologia/@-25.4354389,-49.2779048,17z/data=%213m1%214b1%214m2%213m1%211s0x94dce473a24cf705:0x369fdf05247b568b?hl=pt-BR>
  |  email
<mailto:cont...@wsu.com.br>*
.



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Re: [OpenSIPS-Users] Error in compile of tls_mgm

[Mike Tesliuk]

Hello Razvan,


ii  libssl-dev:amd64  1.1.0f-3  
amd64Secure Sockets Layer toolkit - development files


is the version from the repository, is a clean installation


PS: im sending again cause i send directly to you Razvan not to the list .


Em 21/07/17 11:09, Răzvan Crainea escreveu:
> Hello, Mike!
>
> OpenSIPS tls module uses libssl. can you tell us what version you are
> using for libssl-dev?
>
> Thanks,
> Răzvan Crainea
> OpenSIPS Solutions
> www.opensips-solutions.com
> On 07/21/2017 04:43 PM, Mike Tesliuk wrote:
>> Hello,
>>
>>
>> Im getting this error (below) when trying to compile the tls_mgm on
>> version 2.2.5 (github 2.2)
>>
>>
>> In file included from proto_tls.c:67:0:
>> ../tls_mgm/tls_conn_ops.h: In function ‘tls_conn_init’:
>> ../tls_mgm/tls_conn_ops.h:120:29: error: dereferencing pointer to
>> incomplete type ‘SSL {aka struct ssl_st}’
>>   if ( ((SSL *)c->extra_data)->kssl_ctx ) {
>>  ^~
>> ../tls_mgm/tls_conn_ops.h:121:3: warning: implicit declaration of
>> function ‘kssl_ctx_free’ [-Wimplicit-function-declaration]
>>kssl_ctx_free( ((SSL *)c->extra_data)->kssl_ctx );
>>^
>>
>>
>> The system is a debian 9 with the libs below for tls:
>>
>> libcurl3-gnutls:amd64 - 7.52.1-5
>> libcurl4-gnutls-dev:amd64 - 7.52.1-5
>> libgnutls-dane0:amd64 - 3.5.8-5+deb9u1
>> libgnutls-openssl27:amd64 - 3.5.8-5+deb9u1
>> libgnutls28-dev:amd64 - 3.5.8-5+deb9u1
>> libgnutls30:amd64 - 3.5.8-5+deb9u1
>> libgnutlsxx28:amd64 - 3.5.8-5+deb9u1
>>
>>
>> Can you confirm how can i solve this ?
>>
>> Thank you.
>>
>>
>>
>>
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​Atenciosamente,
WSU TECNOLOGIA
Mike Tesliuk
Rua Visconde do Rio Branco 1630 . Sala 1302 . Curitiba . PR  
12387 SW 125th ter, Miami, Florida 33186 - USA
tel +55 (41) 3941.0650   +1 (786) 719.6253
*website <http://www.wsu.com.br/>  |  mapa
<https://www.google.com.br/maps/place/WSU+Tecnologia/@-25.4354389,-49.2779048,17z/data=%213m1%214b1%214m2%213m1%211s0x94dce473a24cf705:0x369fdf05247b568b?hl=pt-BR>
  |  email
<mailto:cont...@wsu.com.br>*
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[OpenSIPS-Users] Error in compile of tls_mgm

[Mike Tesliuk]

Hello,


Im getting this error (below) when trying to compile the tls_mgm on
version 2.2.5 (github 2.2)


In file included from proto_tls.c:67:0:
../tls_mgm/tls_conn_ops.h: In function ‘tls_conn_init’:
../tls_mgm/tls_conn_ops.h:120:29: error: dereferencing pointer to
incomplete type ‘SSL {aka struct ssl_st}’
  if ( ((SSL *)c->extra_data)->kssl_ctx ) {
 ^~
../tls_mgm/tls_conn_ops.h:121:3: warning: implicit declaration of
function ‘kssl_ctx_free’ [-Wimplicit-function-declaration]
   kssl_ctx_free( ((SSL *)c->extra_data)->kssl_ctx );
   ^


The system is a debian 9 with the libs below for tls:

libcurl3-gnutls:amd64 - 7.52.1-5
libcurl4-gnutls-dev:amd64 - 7.52.1-5
libgnutls-dane0:amd64 - 3.5.8-5+deb9u1
libgnutls-openssl27:amd64 - 3.5.8-5+deb9u1
libgnutls28-dev:amd64 - 3.5.8-5+deb9u1
libgnutls30:amd64 - 3.5.8-5+deb9u1
libgnutlsxx28:amd64 - 3.5.8-5+deb9u1


Can you confirm how can i solve this ?

Thank you.




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Re: [OpenSIPS-Users] Sip trunking and heartbeat

[Mike Tesliuk]

You needd to configure carp , and your sysctl no bind non local IPS, so you
will opensips with  the floating IP address , and the binary replication ,
i have a third server as database too but you can use a máster máster
replication
And how do you that, i mean what should you configure?how do you use the IP
adress?



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Re: [OpenSIPS-Users] Sip trunking and heartbeat

[Mike Tesliuk]

I have an envitoment where i send calls using IP address to SIP trunk, so i
have a carp sharing an IP address between two servers and binary
replication for dialog, i do not have media just signalling and this work
very well
Em 10/03/2016 07:45, "Francjos" <35...@heb.be> escreveu:

> Hello everyone,
>
> I'm using heartbeat on two opensips servers , one active and the other
> passive. I would like to d this:
> When the one which active does not work, the other which is passive cane
> replace it and continue receiving calls.
> The problem i have is the following:
> how  can i do in order one opensips is connected to the sip trunk, and when
> it geos down, the other opensips have to connect to the sip trunk and
> continue to receive the calls?
>
> Thanks
>
>
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/Sip-trunking-and-heartbeat-tp7601972.html
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Re: [OpenSIPS-Users] Opensips + Loadbalancer (inviting himself)

[Mike Tesliuk]

Hello Bogdan,

You are right, i actually dont call the function on a conditional path, so,
probably was because there is no more resources available, i realy dont
remember why i comment out the IF , but i remove the comment and im testing
again

Thanks for the clarification

Have a good day

2016-02-05 7:43 GMT-02:00 Bogdan-Andrei Iancu :

> Hi Mike,
>
> Assuming you do not actually want to have calls going back to your LB, my
> question is : when doing load_balance() function, do you check the return
> code to be sure the function was with success and a new destination was set
> to the call ? maybe the function fails for whatever reason and RURI/DURI is
> untouched, and of course, you route to yourself.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 04.02.2016 19:51, Mike Tesliuk wrote:
>
> Hello Guys,
>
> Im getting a problem with my proxy + loadbalancer.
>
> Some time when i receive an invite the proxy send a new invite to himself
> and so got a authorization request reply
>
> Below you have a link with an image of sngrep showing the situation
>
> http://picpaste.com/pics/abfjhach-gAKIlkfK.1454608071.png
>
> This does not happen all the time with all calls, its a sporadically .
>
> If somebody have an idea about what can generate this kind of issue i will
> appreciate.
>
> Thank you
>
>
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[OpenSIPS-Users] Opensips + Loadbalancer (inviting himself)

[Mike Tesliuk]

Hello Guys,

Im getting a problem with my proxy + loadbalancer.

Some time when i receive an invite the proxy send a new invite to himself
and so got a authorization request reply

Below you have a link with an image of sngrep showing the situation

http://picpaste.com/pics/abfjhach-gAKIlkfK.1454608071.png

This does not happen all the time with all calls, its a sporadically .

If somebody have an idea about what can generate this kind of issue i will
appreciate.

Thank you
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[OpenSIPS-Users] mi_xmlrpc and dlg_list

[Mike Tesliuk]

Hello everybody,


Im trying some tests on an opensips 1.11 [ (1.11.3-notls (x86_64/linux)) 
] using mi_xmlrpc, when i try to call the dlg_list command i get the 
message below






faultCode
-500
faultString
The xmlrpc request could not be parsed into a MI 
tree!






My script is very simple, on the first moment just to check how i can 
receive the return to be parsed


$request = xmlrpc_encode_request("dlg_list","");


$context = stream_context_create(array('http' => array(
'method' => "POST",
'header' => "Content-Type: text/xml",
'content' => $request
)));
$file = file_get_contents("http://127.0.0.1:8000/RPC2";, false, $context);

print $file;
die();


Im able to use other commands like which, get_statistics but not the 
dlg_list, is that expected ?


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Re: [OpenSIPS-Users] LoadBalance - Share resource count between group

[Mike Tesliuk]

Ok bogdan i will do that, thank you

Sent from my iPhone

> On Jul 6, 2015, at 14:09, Bogdan-Andrei Iancu  wrote:
> 
> Hi Mike,
> 
> So answer is no, they do no share - in the profile name we get the 
> destination ID (which is unique) and not the destination IP.
> IF you want, you can open a feature request on github tracker.
> 
> Best regards,
> 
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> 
>> On 03.07.2015 14:02, Mike Tesliuk wrote:
>> 
>> [root@sipproxy1-R3 ~]# opensipsctl fifo list_all_profiles
>> lbX41:: 1
>> lbX31:: 1
>> lbX21:: 1
>> lbX20:: 1
>> 
>> 
>>> On 03/07/15 05:09, Bogdan-Andrei Iancu wrote:
>>> Hi Mike,
>>> 
>>> Could you do a list_all_profiles MI command:
>>> http://www.opensips.org/html/docs/modules/1.11.x/dialog.html#id297069
>>> 
>>> Regards,
>>> 
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developer
>>> http://www.opensips-solutions.com
>>> 
>>>> On 03.07.2015 06:10, Mike Tesliuk wrote:
>>>> 
>>>> Hello Bogdan,
>>>> 
>>>> seems that the share is not working as you can see below
>>>> 
>>>> Destination:: sip:177.43.X07.XX6:5060 id=33 group=2 enabled=yes auto-re=on
>>>>Resource:: 41 max=42 load=0
>>>>Resource:: 31 max=75 load=0
>>>>Resource:: 21 max=64 load=0
>>>>Resource:: 20 max=112 load=0
>>>> Destination:: sip:177.43.X07.XX6:5060 id=13 group=1 enabled=yes auto-re=on
>>>>Resource:: 41 max=42 load=1
>>>>Resource:: 31 max=75 load=0
>>>>Resource:: 21 max=64 load=0
>>>>Resource:: 20 max=112 load=0
>>>> 
>>>> 
>>>> i try to add the /s on the end of the resource name (41 is a gsm carrier 
>>>> identification, so i try 41/s=42 ) but when i add this the gateways dos 
>>>> not show on lb_list anymore.
>>>> 
>>>> Im using on this gateway an old 1.9 version [ Server:: OpenSIPS 
>>>> (1.9.1-notls (x86_64/linux)) ]
>>>> 
>>>> i can update if neede as this is a new equipment, but i had made this 
>>>> instalation a year ago, so this is why im using version 1.9
>>>> 
>>>> Thank you.
>>>> 
>>>> 
>>>> 
>>>>> On 02/07/15 10:47, Bogdan-Andrei Iancu wrote:
>>>>> Hi Mike,
>>>>> 
>>>>> 
>>>>> The dialog profiles (used for counting the calls) are internally build 
>>>>> using the resource name and the IP of the destination. So, if same 
>>>>> destination (as IP) , same resource name, in different rows, for 
>>>>> different groups, the same profile should be used in both cases (it will 
>>>>> be shared). I say "should" as I never tried this case, but looking at the 
>>>>> code it should share the load.
>>>>> Just give it a try and let me know ;)
>>>>> 
>>>>> Best regards,
>>>>> 
>>>>> Bogdan-Andrei Iancu
>>>>> OpenSIPS Founder and Developer
>>>>> http://www.opensips-solutions.com
>>>>> 
>>>>>> On 02.07.2015 02:57, Mike Tesliuk wrote:
>>>>>> Hello all,
>>>>>> 
>>>>>> Is that possible to share the count between two diferent groups on 
>>>>>> loadbalance ?
>>>>>> 
>>>>>> I will have the same sip-uri but i need diferent groups because i need 
>>>>>> to use diferent gateways according with the originator, one originator 
>>>>>> can have two gateways, the other can have just one, but the resources 
>>>>>> need to be visible between groups.
>>>>>> 
>>>>>> Is that possible ?
>>>>>> 
>>>>>> Thank you.
>>>>>> 
>>>>>> ___
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>>>>>> Users@lists.opensips.org
>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 

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Re: [OpenSIPS-Users] LoadBalance - Share resource count between group

[Mike Tesliuk]

[root@sipproxy1-R3 ~]# opensipsctl fifo list_all_profiles
lbX41:: 1
lbX31:: 1
lbX21:: 1
lbX20:: 1


On 03/07/15 05:09, Bogdan-Andrei Iancu wrote:

Hi Mike,

Could you do a list_all_profiles MI command:
http://www.opensips.org/html/docs/modules/1.11.x/dialog.html#id297069

Regards,

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

On 03.07.2015 06:10, Mike Tesliuk wrote:


Hello Bogdan,

seems that the share is not working as you can see below

Destination:: sip:177.43.X07.XX6:5060 id=33 group=2 enabled=yes 
auto-re=on

Resource:: 41 max=42 load=0
Resource:: 31 max=75 load=0
Resource:: 21 max=64 load=0
Resource:: 20 max=112 load=0
Destination:: sip:177.43.X07.XX6:5060 id=13 group=1 enabled=yes 
auto-re=on

Resource:: 41 max=42 load=1
Resource:: 31 max=75 load=0
Resource:: 21 max=64 load=0
Resource:: 20 max=112 load=0


i try to add the /s on the end of the resource name (41 is a gsm 
carrier identification, so i try 41/s=42 ) but when i add this the 
gateways dos not show on lb_list anymore.


Im using on this gateway an old 1.9 version [ Server:: OpenSIPS 
(1.9.1-notls (x86_64/linux)) ]


i can update if neede as this is a new equipment, but i had made this 
instalation a year ago, so this is why im using version 1.9


Thank you.



On 02/07/15 10:47, Bogdan-Andrei Iancu wrote:

Hi Mike,


The dialog profiles (used for counting the calls) are internally 
build using the resource name and the IP of the destination. So, if 
same destination (as IP) , same resource name, in different rows, 
for different groups, the same profile should be used in both cases 
(it will be shared). I say "should" as I never tried this case, but 
looking at the code it should share the load.

Just give it a try and let me know ;)

Best regards,

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

On 02.07.2015 02:57, Mike Tesliuk wrote:

Hello all,

Is that possible to share the count between two diferent groups on 
loadbalance ?


I will have the same sip-uri but i need diferent groups because i 
need to use diferent gateways according with the originator, one 
originator can have two gateways, the other can have just one, but 
the resources need to be visible between groups.


Is that possible ?

Thank you.

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Re: [OpenSIPS-Users] LoadBalance - Share resource count between group

[Mike Tesliuk]

Hello Bogdan,

seems that the share is not working as you can see below

Destination:: sip:177.43.X07.XX6:5060 id=33 group=2 enabled=yes auto-re=on
Resource:: 41 max=42 load=0
Resource:: 31 max=75 load=0
Resource:: 21 max=64 load=0
Resource:: 20 max=112 load=0
Destination:: sip:177.43.X07.XX6:5060 id=13 group=1 enabled=yes auto-re=on
Resource:: 41 max=42 load=1
Resource:: 31 max=75 load=0
Resource:: 21 max=64 load=0
Resource:: 20 max=112 load=0


i try to add the /s on the end of the resource name (41 is a gsm carrier 
identification, so i try 41/s=42 ) but when i add this the gateways dos 
not show on lb_list anymore.


Im using on this gateway an old 1.9 version [ Server:: OpenSIPS 
(1.9.1-notls (x86_64/linux)) ]


i can update if neede as this is a new equipment, but i had made this 
instalation a year ago, so this is why im using version 1.9


Thank you.



On 02/07/15 10:47, Bogdan-Andrei Iancu wrote:

Hi Mike,


The dialog profiles (used for counting the calls) are internally build 
using the resource name and the IP of the destination. So, if same 
destination (as IP) , same resource name, in different rows, for 
different groups, the same profile should be used in both cases (it 
will be shared). I say "should" as I never tried this case, but 
looking at the code it should share the load.

Just give it a try and let me know ;)

Best regards,

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

On 02.07.2015 02:57, Mike Tesliuk wrote:

Hello all,

Is that possible to share the count between two diferent groups on 
loadbalance ?


I will have the same sip-uri but i need diferent groups because i 
need to use diferent gateways according with the originator, one 
originator can have two gateways, the other can have just one, but 
the resources need to be visible between groups.


Is that possible ?

Thank you.

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[OpenSIPS-Users] LoadBalance - Share resource count between group

[Mike Tesliuk]

Hello all,

Is that possible to share the count between two diferent groups on 
loadbalance ?


I will have the same sip-uri but i need diferent groups because i need 
to use diferent gateways according with the originator, one originator 
can have two gateways, the other can have just one, but the resources 
need to be visible between groups.


Is that possible ?

Thank you.

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Re: [OpenSIPS-Users] Opensips stop processing

[Mike Tesliuk]

ok, i will check this, and tell what happen, thanks


On 20/02/15 11:21, Liviu Chircu wrote:

Hello Mike,

Critical things to watch out for:
* CPU : if one or more opensips processes are stuck in 100% CPU, 
try to obtain a backtrace with "opensipsctl trap"
* insufficient shared memory: "opensipsctl fifo get_statistics 
shmem:" - if real_usage / max_real_usage exceed 70% of your 
total_size, increase the "-m" parameter in your initscript!
* UDP/TCP queue: "netstat -ulnp", watch for the "Recv-Q" column of 
your opensips processes. If INVITEs are not being processed at all, 
you should see them there
* debug levels: make sure OpenSIPS reports errors! "opensipsctl 
fifo debug 3"


Best regards,

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

On 20.02.2015 11:49, Mike Tesliuk wrote:

Hello ,

Im getting a strange error where opensips simply stop to reply the 
invites, i have no errors on log the system does not crash, just does 
not reply the invites anymore


Somebody have an idea of how can i debug this ?

Thank you

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[OpenSIPS-Users] Opensips stop processing

[Mike Tesliuk]

Hello ,

Im getting a strange error where opensips simply stop to reply the 
invites, i have no errors on log the system does not crash, just does 
not reply the invites anymore


Somebody have an idea of how can i debug this ?

Thank you

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[OpenSIPS-Users] sip code 487 and log_next_state_dlg: bogus event 4 in state 5

[Mike Tesliuk]

Hello Guys,


im getting this critical message on my opensips, but i see that this 
message just appear with the binary replication activated, is this 
expected ?


Feb 19 07:34:12 sipproxy01 /usr/sbin/opensips[13762]: 
CRITICAL:dialog:log_next_state_dlg: bogus event 4 in state 5 for dlg 
0x7fb6de3922b0 [1887:354100080] with clid 
'37e9a2df45cc258436f4cb351e60832b@172.30.1.15:5060' and tags 
'as33b7ba8a' 'as40351bae'
Feb 19 07:34:12 sipproxy01 /usr/sbin/opensips[13762]: 
DBG:dialog:next_state_dlg: dialog 0x7fb6de3922b0 changed from state 5 to 
state 5, due event 4
Feb 19 07:34:12 sipproxy01 /usr/sbin/opensips[13762]: 
DBG:tm:cleanup_uac_timers: RETR/FR timers reset





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Re: [OpenSIPS-Users] Dailog State 5

[Mike Tesliuk]

Hello,

The firewall on the server was using a default policy to DROP and was 
not configured with an INPUT rule to loopback interface, after add the 
rule the dialogs start to run correctly .


i think that the problem is solved.




On 07/02/15 14:20, Mike Tesliuk wrote:

Hello everybody

Im still with this problem, when i reboot the server (  a VM ) the 
problem disappear but return some time later


in 10 minutes i have like more than 2000 dialogs on state 5

after a restart the state 5 dialogs are clean up one or two times and 
after that just increase


i had compiled again to get the memory memlog running , below some 
information


 opensipsctl fifo uptime
Now:: Sat Feb  7 13:42:08 2015
Up since:: Sat Feb  7 13:37:01 2015
Up time:: 307 [sec]

my dialogs:

  8 state:: 1
104 state:: 2
 68 state:: 4
   2488 state:: 5


Below the link to the memory dump

http://pastebin.com/Lz54EZzH


The problem is that if i dont restart the opensips i will get this 
dialog increasing and i will receive a out of memory


Below you have the dialogs for 20 minutes:

  6 state:: 1
 88 state:: 2
 62 state:: 4
  11253 state:: 5


And on get_statistics fifo command i get this:

dialog:active_dialogs:: 58
dialog:early_dialogs:: 104
dialog:processed_dialogs:: 10604
dialog:expired_dialogs:: 0
dialog:failed_dialogs:: 8726

shmem:total_size:: 2147483648
shmem:used_size:: 313933008
shmem:real_used_size:: 323767368
shmem:max_used_size:: 323768544
shmem:free_size:: 1823716280
shmem:fragments:: 5

Below the pkmem information:

http://pastebin.com/RXKBTqbZ


I will realy apreciate if somebody can help with this.

Thank you.





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Re: [OpenSIPS-Users] Dailog State 5

[Mike Tesliuk]

Hello everybody

Im still with this problem, when i reboot the server (  a VM ) the 
problem disappear but return some time later


in 10 minutes i have like more than 2000 dialogs on state 5

after a restart the state 5 dialogs are clean up one or two times and 
after that just increase


i had compiled again to get the memory memlog running , below some 
information


 opensipsctl fifo uptime
Now:: Sat Feb  7 13:42:08 2015
Up since:: Sat Feb  7 13:37:01 2015
Up time:: 307 [sec]

my dialogs:

  8 state:: 1
104 state:: 2
 68 state:: 4
   2488 state:: 5


Below the link to the memory dump

http://pastebin.com/Lz54EZzH


The problem is that if i dont restart the opensips i will get this 
dialog increasing and i will receive a out of memory


Below you have the dialogs for 20 minutes:

  6 state:: 1
 88 state:: 2
 62 state:: 4
  11253 state:: 5


And on get_statistics fifo command i get this:

dialog:active_dialogs:: 58
dialog:early_dialogs:: 104
dialog:processed_dialogs:: 10604
dialog:expired_dialogs:: 0
dialog:failed_dialogs:: 8726

shmem:total_size:: 2147483648
shmem:used_size:: 313933008
shmem:real_used_size:: 323767368
shmem:max_used_size:: 323768544
shmem:free_size:: 1823716280
shmem:fragments:: 5

Below the pkmem information:

http://pastebin.com/RXKBTqbZ


I will realy apreciate if somebody can help with this.

Thank you.


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Re: [OpenSIPS-Users] Dailog State 5

[Mike Tesliuk]

Hello Liviu,

The problem disappear after a reboot on the server , now the dialogs are 
being removed well.


Thanks for your help

Em 10/12/14 15:34, Mike Tesliuk escreveu:

Hello Liviu,

I had change the db_mode to 2  (was using this before) and change the 
db_update_period to 60, but this does not solve the problem.


below you have a link to paste bin where you can find the trace and 
the dialog information.


http://pastebin.com/tWF6JWRr

I hope you can give me some information.

Thank you very much for your colaboration until now.






Em 10/12/14 13:19, Liviu Chircu escreveu:

I'm going to ask you to try/do the following things:

* fiddle with the "db_update_period" parameter of the dialog module 
[1]. See what works best for you.
With db_mode=1, it's normal to have more dialogs in memory than in 
the database at a given time. Please remember

that all dialogs are immediately flushed to the DB at shutdown anyway.

* If you suspect that dialogs are really "hanging" in state 5, and 
not being removed, could you try to extract a
SIP trace for such a problematic call, so we can try and replicate 
the problem ourselves? Thank you!


[1]: 
http://www.opensips.org/html/docs/modules/2.1.x/dialog.html#id294076


Best regards,

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

On 10.12.2014 15:57, Mike Tesliuk wrote:


Hello Liviu,

Thanks for you answer, so, the opensips is starting with -m 3072 and 
-M 256


the get_statistics shmem is below (but i had executed a restart 5 
minutes ago)


shmem:total_size:: 3221225472
shmem:used_size:: 53679760
shmem:real_used_size:: 55528624
shmem:max_used_size:: 55528664
shmem:free_size:: 3165696848
shmem:fragments:: 2

10 seconds later


shmem:total_size:: 3221225472
shmem:used_size:: 56895544
shmem:real_used_size:: 58833520
shmem:max_used_size:: 58833520
shmem:free_size:: 3162391952
shmem:fragments:: 2


we have now 3478 dialogs on memory which 2061 are state 5



Dec 10 11:54:01 sipproxy2 /usr/local/opensips/sbin/opensips[22057]: 
DBG:dialog:dlg_onroute: route param is 'e28.d3c54054' (len=12)
Dec 10 11:54:01 sipproxy2 /usr/local/opensips/sbin/opensips[22057]: 
DBG:dialog:lookup_dlg: no dialog id=1157913661 found on entry 2094
Dec 10 11:54:01 sipproxy2 /usr/local/opensips/sbin/opensips[22057]: 
DBG:dialog:dlg_onroute: unable to find dialog for BYE with route 
param 'e28.d3c54054'
Dec 10 11:54:01 sipproxy2 /usr/local/opensips/sbin/opensips[22057]: 
DBG:dialog:get_dlg: input ci=(30), 
tt=(36), ft=(10)
Dec 10 11:54:01 sipproxy2 /usr/local/opensips/sbin/opensips[22057]: 
DBG:dialog:get_dlg: no dialog 
callid='c36ef238f2382b05@66.77.199.182' found
Dec 10 11:54:01 sipproxy2 /usr/local/opensips/sbin/opensips[22057]: 
DBG:dialog:dlg_onroute: Callid 'c36ef238f2382b05@66.77.199.182' not 
found



and on database, i have

mysql> select count(*) from dialog;
+--+
| count(*) |
+--+
| 1465 |
+--+

(now while i finish to write this email i have 3703 dialogs on state 5)



Thanks in advice for you help.



Em 10/12/14 10:24, Liviu Chircu escreveu:

Hello Mike,

Normally, state 5 means that the dialog is over, and just waiting 
to be deleted from memory (BYEs have been sent!).


From those numbers, it looks like you have enough shared memory, 
but no sufficient pkg memory.

Could you please provide the following:

* amount of SHM and PKG you start OpenSIPS with
* any relevant ERRORs you see in the logfile
* output of "opensipsctl fifo get_statistics shmem:" command when 
you start to see memory errors


Best regards,

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

On 10.12.2014 14:13, Mike Tesliuk wrote:

Hello Guys,

Im getting a problem of out of memory on my opensips, and what i 
see is that i have a lot more dialogs on memory then on database, 
im using db_mode on 1 , and i have now on database 1396 dialogs 
and 35823 on memory, on memmory a lot of them are on state 5.


What can cause this problem ? how can i identify this ?

Thank you

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Re: [OpenSIPS-Users] Dailog State 5

[Mike Tesliuk]

Hello Liviu,

I had change the db_mode to 2  (was using this before) and change the 
db_update_period to 60, but this does not solve the problem.


below you have a link to paste bin where you can find the trace and the 
dialog information.


http://pastebin.com/tWF6JWRr

I hope you can give me some information.

Thank you very much for your colaboration until now.






Em 10/12/14 13:19, Liviu Chircu escreveu:

I'm going to ask you to try/do the following things:

* fiddle with the "db_update_period" parameter of the dialog module 
[1]. See what works best for you.
With db_mode=1, it's normal to have more dialogs in memory than in the 
database at a given time. Please remember

that all dialogs are immediately flushed to the DB at shutdown anyway.

* If you suspect that dialogs are really "hanging" in state 5, and not 
being removed, could you try to extract a
SIP trace for such a problematic call, so we can try and replicate the 
problem ourselves? Thank you!


[1]: http://www.opensips.org/html/docs/modules/2.1.x/dialog.html#id294076

Best regards,

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

On 10.12.2014 15:57, Mike Tesliuk wrote:


Hello Liviu,

Thanks for you answer, so, the opensips is starting with -m 3072 and 
-M 256


the get_statistics shmem is below (but i had executed a restart 5 
minutes ago)


shmem:total_size:: 3221225472
shmem:used_size:: 53679760
shmem:real_used_size:: 55528624
shmem:max_used_size:: 55528664
shmem:free_size:: 3165696848
shmem:fragments:: 2

10 seconds later


shmem:total_size:: 3221225472
shmem:used_size:: 56895544
shmem:real_used_size:: 58833520
shmem:max_used_size:: 58833520
shmem:free_size:: 3162391952
shmem:fragments:: 2


we have now 3478 dialogs on memory which 2061 are state 5



Dec 10 11:54:01 sipproxy2 /usr/local/opensips/sbin/opensips[22057]: 
DBG:dialog:dlg_onroute: route param is 'e28.d3c54054' (len=12)
Dec 10 11:54:01 sipproxy2 /usr/local/opensips/sbin/opensips[22057]: 
DBG:dialog:lookup_dlg: no dialog id=1157913661 found on entry 2094
Dec 10 11:54:01 sipproxy2 /usr/local/opensips/sbin/opensips[22057]: 
DBG:dialog:dlg_onroute: unable to find dialog for BYE with route 
param 'e28.d3c54054'
Dec 10 11:54:01 sipproxy2 /usr/local/opensips/sbin/opensips[22057]: 
DBG:dialog:get_dlg: input ci=(30), 
tt=(36), ft=(10)
Dec 10 11:54:01 sipproxy2 /usr/local/opensips/sbin/opensips[22057]: 
DBG:dialog:get_dlg: no dialog callid='c36ef238f2382b05@66.77.199.182' 
found
Dec 10 11:54:01 sipproxy2 /usr/local/opensips/sbin/opensips[22057]: 
DBG:dialog:dlg_onroute: Callid 'c36ef238f2382b05@66.77.199.182' not 
found



and on database, i have

mysql> select count(*) from dialog;
+--+
| count(*) |
+--+
| 1465 |
+--+

(now while i finish to write this email i have 3703 dialogs on state 5)



Thanks in advice for you help.



Em 10/12/14 10:24, Liviu Chircu escreveu:

Hello Mike,

Normally, state 5 means that the dialog is over, and just waiting to 
be deleted from memory (BYEs have been sent!).


From those numbers, it looks like you have enough shared memory, but 
no sufficient pkg memory.

Could you please provide the following:

* amount of SHM and PKG you start OpenSIPS with
* any relevant ERRORs you see in the logfile
* output of "opensipsctl fifo get_statistics shmem:" command when 
you start to see memory errors


Best regards,

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

On 10.12.2014 14:13, Mike Tesliuk wrote:

Hello Guys,

Im getting a problem of out of memory on my opensips, and what i 
see is that i have a lot more dialogs on memory then on database, 
im using db_mode on 1 , and i have now on database 1396 dialogs and 
35823 on memory, on memmory a lot of them are on state 5.


What can cause this problem ? how can i identify this ?

Thank you

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Re: [OpenSIPS-Users] Dailog State 5

[Mike Tesliuk]

Hello Liviu,

Thanks for you answer, so, the opensips is starting with -m 3072 and -M 256

the get_statistics shmem is below (but i had executed a restart 5 
minutes ago)


shmem:total_size:: 3221225472
shmem:used_size:: 53679760
shmem:real_used_size:: 55528624
shmem:max_used_size:: 55528664
shmem:free_size:: 3165696848
shmem:fragments:: 2

10 seconds later


shmem:total_size:: 3221225472
shmem:used_size:: 56895544
shmem:real_used_size:: 58833520
shmem:max_used_size:: 58833520
shmem:free_size:: 3162391952
shmem:fragments:: 2


we have now 3478 dialogs on memory which 2061 are state 5



Dec 10 11:54:01 sipproxy2 /usr/local/opensips/sbin/opensips[22057]: 
DBG:dialog:dlg_onroute: route param is 'e28.d3c54054' (len=12)
Dec 10 11:54:01 sipproxy2 /usr/local/opensips/sbin/opensips[22057]: 
DBG:dialog:lookup_dlg: no dialog id=1157913661 found on entry 2094
Dec 10 11:54:01 sipproxy2 /usr/local/opensips/sbin/opensips[22057]: 
DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 
'e28.d3c54054'
Dec 10 11:54:01 sipproxy2 /usr/local/opensips/sbin/opensips[22057]: 
DBG:dialog:get_dlg: input ci=(30), 
tt=(36), ft=(10)
Dec 10 11:54:01 sipproxy2 /usr/local/opensips/sbin/opensips[22057]: 
DBG:dialog:get_dlg: no dialog callid='c36ef238f2382b05@66.77.199.182' found
Dec 10 11:54:01 sipproxy2 /usr/local/opensips/sbin/opensips[22057]: 
DBG:dialog:dlg_onroute: Callid 'c36ef238f2382b05@66.77.199.182' not found



and on database, i have

mysql> select count(*) from dialog;
+--+
| count(*) |
+--+
| 1465 |
+--+

(now while i finish to write this email i have 3703 dialogs on state 5)



Thanks in advice for you help.



Em 10/12/14 10:24, Liviu Chircu escreveu:

Hello Mike,

Normally, state 5 means that the dialog is over, and just waiting to 
be deleted from memory (BYEs have been sent!).


From those numbers, it looks like you have enough shared memory, but 
no sufficient pkg memory.

Could you please provide the following:

* amount of SHM and PKG you start OpenSIPS with
* any relevant ERRORs you see in the logfile
* output of "opensipsctl fifo get_statistics shmem:" command when you 
start to see memory errors


Best regards,

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

On 10.12.2014 14:13, Mike Tesliuk wrote:

Hello Guys,

Im getting a problem of out of memory on my opensips, and what i see 
is that i have a lot more dialogs on memory then on database, im 
using db_mode on 1 , and i have now on database 1396 dialogs and 
35823 on memory, on memmory a lot of them are on state 5.


What can cause this problem ? how can i identify this ?

Thank you

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[OpenSIPS-Users] Dailog State 5

[Mike Tesliuk]

Hello Guys,

Im getting a problem of out of memory on my opensips, and what i see is 
that i have a lot more dialogs on memory then on database, im using 
db_mode on 1 , and i have now on database 1396 dialogs and 35823 on 
memory, on memmory a lot of them are on state 5.


What can cause this problem ? how can i identify this ?

Thank you

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[OpenSIPS-Users] load_balance profile failover

[Mike Tesliuk]

Hello Guys,

Im trying to implement a diferent feature using load balance, i check on 
maillist archive and check on this post  ( 
http://www.openser.org/pipermail/users/2013-July/026269.html ) that we 
must use the same resource on load balance, when i try a diferent way i 
got this message


Nov 19 17:58:39 sipproxy01 /usr/sbin/opensips[32010]: 
ERROR:load_balancer:do_load_balance: failed to remove from profile

Nov 19 17:58:39 sipproxy01 /usr/sbin/opensips[32010]: Operadora: algar
Nov 19 17:58:39 sipproxy01 /usr/sbin/opensips[32010]: 
ERROR:load_balancer:do_load_balance: failed to remove from profile

Nov 19 17:58:39 sipproxy01 /usr/sbin/opensips[32010]: Operadora: oi
Nov 19 17:58:39 sipproxy01 /usr/sbin/opensips[32010]: 
ERROR:load_balancer:do_load_balance: failed to remove from profile

Nov 19 17:58:39 sipproxy01 /usr/sbin/opensips[32010]: Operadora: ebt
Nov 19 17:58:39 sipproxy01 /usr/sbin/opensips[32010]: 
ERROR:load_balancer:do_load_balance: failed to remove from profile



So, what i need to do is to check the kind of route (landline, mobile 
numbers, long distance, service number etc..) and with that information 
i use an array to choose the resource that i need on loadbalance, if 
that resource fail, i need to send to another group of resource, let my 
put a sample here:


if i got a call for local mobile number i create this array:

case "VC1":
# Variaveis devem ser adicionadas em ordem inversa
$avp(max_operadora) = "7";
$avp(ordem_operadora) = "gvt";
$avp(ordem_operadora) = "ebt";
$avp(ordem_operadora) = "oi";
$avp(ordem_operadora) = "algar";
$avp(ordem_operadora) = "transit";
$avp(ordem_operadora) = "yama";
$avp(ordem_operadora) = "njgsm";
$avp(ordem_operadora) = "claro";
break;

and so, i send to the load balancer

if( !load_balance("10","$(avp(ordem_operadora)[$avp(contador)])")){

if i got a fail i will loop on loadbalance again and change the counter 
, so i will use the next carrier


If i cannot implement this on load_balance, which module should i use ?

If im not wrong with drouting i cannot create a load_balance right ?

The dispatcher module seems to be more accurate for this function, 
somebody can confirm that ?


Thank you guys.



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Re: [OpenSIPS-Users] how to log out opensips

[Mike Tesliuk]

Take a look here

http://www.opensips.org/html/docs/modules/devel/registrar.html#id293748


You can define how many contact a user can have , so, if you put 1
just the last registration will be used.



2014-05-08 8:09 GMT-03:00 老鬼 :
> I get a linphone client for my phone and construct a opensips servers on my
> Ubuntu computer. I register my linphone client on the opensips servers by
> user name 101, and register user name 102 for another phone , and establish
> a video call between these two users successfully. But when I replace the
> user 101 with 103, and use 102 to call 101 it was also call the phone that
> register by user 103. How to unregister the user ? It's there a command to
> unregister a user?Please help me, thank you .
>
>
>
>
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Re: [OpenSIPS-Users] Database call

[Mike Tesliuk]

You can use the avp_db_query for this operation.

http://www.opensips.org/html/docs/modules/1.10.x/avpops.html#id293988

2014-04-29 5:55 GMT-03:00 Mike Claudi Pedersen :
> Is there someway to refer to a value from the location table, like making a
> database call, i want to do this in some special cases where the called
> number is a shortnumber and then refer to which comapny the shortnumber is
> in... the company values are in my database. but i have no idea have to
> refer to them?
>
> any help would be greatly appreciated
>
>
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Re: [OpenSIPS-Users] Opensips tech prefix

[Mike Tesliuk]

Im using the the techprefix using the contextinfo from address table,
work very well, and for users that whant to authenticate i use the
techprefix information from usr_preferences table using avp_db_query
to fetch the value

2014-04-02 5:33 GMT-03:00 Bogdan-Andrei Iancu :
> Hello,
>
> There is no other way but to use the aaa_radius module with the
> radius_send_auth() (
> http://www.opensips.org/html/docs/modules/1.11.x/aaa_radius.html#id293422)
> and use custom input sets of RADIUS AVPs
> (http://www.opensips.org/html/docs/modules/1.11.x/aaa_radius.html#id249603)
> to be able to push the tech prefix to the RADIUS server.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 10.03.2014 18:54, Alcindo Schleder wrote:
>
> How to authenticate customers without username and password using tech
> prefix on a AAA server and opensips?
>
>
>
> Alcindo Schleder
>
>
>
>
>
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>
>
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Re: [OpenSIPS-Users] Bad from on header

[Mike Tesliuk]

Ok, i will try this , thank you


2014-03-17 11:31 GMT-04:00 Bogdan-Andrei Iancu :

>  Yhe ACK matches the dialog, but the UAC module is not able to do a
> correct restore / change of the FROM header.
>
> BTW, try to do the FROM stuff after creating the dialog -> in this case,
> the B64 value will be stored in the dialog and not in the RR header.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 17.03.2014 17:17, Mike Tesliuk wrote:
>
>  Bogdan,
>
>  A last question about that, in this case, the ack does not match, so, the
> dialog still in the memory right  ? will not be ended , and this can be the
> problem that i have with memory on this server (too high memory usage)
>
>
> 2014-03-17 11:03 GMT-04:00 Mike Tesliuk :
>
>> Ok Bogdan, thanks for you explanation, i will check that
>>
>>
>> 2014-03-17 10:43 GMT-04:00 Bogdan-Andrei Iancu :
>>
>>  Yes, definitely that is the problem - that string is a B64 encoded
>>> value, so it is case sensitive. According to RFC3261, UAs must copy the RR
>>> params without any change (even if they do not understand). So your UAC is
>>> broken when comes to handling RR headers.
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>
>>>   On 17.03.2014 15:36, Mike Tesliuk wrote:
>>>
>>> Ok,
>>>
>>>  I receive the invite, and i send the invite with this vsf
>>> AF5CW0NKbgEAcAR4BxYLHgAABB0FGgA4
>>>
>>>  on the 180 is ok
>>>  on the 183 is ok
>>>  on the 200 is ok
>>>
>>>  The ack is ok, but lower case ( af5cw0nkbgeacar4bxylhgaabb0fgga4 )
>>>
>>>  so everything broke
>>>
>>> From: <...S.m.z._6.A..^$..4.X.-.5.>;tag=1c731545057.
>>>
>>>
>>>  The lower case on ACK can be the problem ?
>>>
>>>
>>>
>>> 2014-03-17 5:28 GMT-04:00 Bogdan-Andrei Iancu :
>>>
>>>>  Hello Mike,
>>>>
>>>> The restore/change of the FROM hdr is done automatically for the
>>>> sequential requests. What I suspect in your case is an altering of the
>>>> RR/Route "vsf" param - check if you have the same value in the outgoing
>>>> INVITE (original), in the invite 200 OK (in RR hdr) and the incoming ACK
>>>> (in Route hdr)
>>>>
>>>> Regards,
>>>>
>>>> Bogdan-Andrei Iancu
>>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>>
>>>>   On 17.03.2014 00:03, Mike Tesliuk wrote:
>>>>
>>>>  Hello Bogdan,
>>>>
>>>>  Yes, on the initial request we have the uac_replace_from , but in this
>>>> case the ack is not supposed to reach the function , as i say before, this
>>>> does not happen on every dialog, just in some situation that i dont
>>>> identify exactly which one yet.
>>>>
>>>>
>>>> 2014-03-16 17:06 GMT-04:00 Bogdan-Andrei Iancu :
>>>>
>>>>>  Hello Mike,
>>>>>
>>>>> Are you using the uac_replace_from() for that call ?
>>>>>
>>>>> Regards,
>>>>>
>>>>> Bogdan-Andrei Iancu
>>>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>>>
>>>>>  On 14.03.2014 19:48, Mike Tesliuk wrote:
>>>>>
>>>>> Hello Guys,
>>>>>
>>>>> Im checking about a problem here i hope somebody can help me.
>>>>>
>>>>>  Some times when i receive an ACK opensips is doing something strange
>>>>> with the from, check the mesage below
>>>>>
>>>>>  Message that opensips has received
>>>>>
>>>>> U _CUSTOMER_IP_:5060 -> __OPENSIPS_IP__:5060
>>>>> ACK sip:603#558533822977@_ASTERISK_IP_:5060 SIP/2.0.
>>>>> Contact: .
>>>>> CSeq: 1 ACK.
>>>>> From: ;tag=1c512295868.
>>>>>
>>>>>  Message that opensips has sended
>>>>>
>>>>> U __OPENSIPS_IP__:5060 -> _ASTERISK_IP_:5060
>>>>> ACK sip:603#558533822977@_ASTERISK_IP_:5060 SIP/2.0.
>>>>> Contact: .
>>>>> CSeq: 1 ACK.
>>>>> From: <...S.i.~.\..)..N(..24Y3-.9.>;tag=1c512295868.
>>>>>
>>>>>  With this, i get on my log  messages like below.
>>>>>
>>>>> ERROR:core:parse_to: unexpected char [\] in status 6:
>>>>> <<<#032??S?i?~?>> .
>>>>> ERROR:core:parse_from_header: bad from header
>>>>>  ERROR:uac:restore_uris_reply: failed to find/parse FROM hdr
>>>>>
>>>>>  and the acc show me this
>>>>>
>>>>> reason=Call leg/transaction does not exist
>>>>>
>>>>>  i dont understand why this is happen, but happen just some times, and
>>>>> as i can find just with one customer.
>>>>>
>>>>>
>>>>>  if somebody can point me what kind of mistake can generate this error
>>>>> i will apreciate.
>>>>>
>>>>>  Thanks
>>>>>
>>>>>
>>>>>  ___
>>>>> Users mailing 
>>>>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>>
>>>>
>>>>
>>>
>>>
>>
>
>
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Re: [OpenSIPS-Users] Bad from on header

[Mike Tesliuk]

Bogdan,

A last question about that, in this case, the ack does not match, so, the
dialog still in the memory right  ? will not be ended , and this can be the
problem that i have with memory on this server (too high memory usage)


2014-03-17 11:03 GMT-04:00 Mike Tesliuk :

> Ok Bogdan, thanks for you explanation, i will check that
>
>
> 2014-03-17 10:43 GMT-04:00 Bogdan-Andrei Iancu :
>
>  Yes, definitely that is the problem - that string is a B64 encoded
>> value, so it is case sensitive. According to RFC3261, UAs must copy the RR
>> params without any change (even if they do not understand). So your UAC is
>> broken when comes to handling RR headers.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 17.03.2014 15:36, Mike Tesliuk wrote:
>>
>> Ok,
>>
>>  I receive the invite, and i send the invite with this vsf
>> AF5CW0NKbgEAcAR4BxYLHgAABB0FGgA4
>>
>>  on the 180 is ok
>>  on the 183 is ok
>>  on the 200 is ok
>>
>>  The ack is ok, but lower case ( af5cw0nkbgeacar4bxylhgaabb0fgga4 )
>>
>>  so everything broke
>>
>> From: <...S.m.z._6.A..^$..4.X.-.5.>;tag=1c731545057.
>>
>>
>>  The lower case on ACK can be the problem ?
>>
>>
>>
>> 2014-03-17 5:28 GMT-04:00 Bogdan-Andrei Iancu :
>>
>>>  Hello Mike,
>>>
>>> The restore/change of the FROM hdr is done automatically for the
>>> sequential requests. What I suspect in your case is an altering of the
>>> RR/Route "vsf" param - check if you have the same value in the outgoing
>>> INVITE (original), in the invite 200 OK (in RR hdr) and the incoming ACK
>>> (in Route hdr)
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>
>>>   On 17.03.2014 00:03, Mike Tesliuk wrote:
>>>
>>>  Hello Bogdan,
>>>
>>>  Yes, on the initial request we have the uac_replace_from , but in this
>>> case the ack is not supposed to reach the function , as i say before, this
>>> does not happen on every dialog, just in some situation that i dont
>>> identify exactly which one yet.
>>>
>>>
>>> 2014-03-16 17:06 GMT-04:00 Bogdan-Andrei Iancu :
>>>
>>>>  Hello Mike,
>>>>
>>>> Are you using the uac_replace_from() for that call ?
>>>>
>>>> Regards,
>>>>
>>>> Bogdan-Andrei Iancu
>>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>>
>>>>  On 14.03.2014 19:48, Mike Tesliuk wrote:
>>>>
>>>> Hello Guys,
>>>>
>>>> Im checking about a problem here i hope somebody can help me.
>>>>
>>>>  Some times when i receive an ACK opensips is doing something strange
>>>> with the from, check the mesage below
>>>>
>>>>  Message that opensips has received
>>>>
>>>> U _CUSTOMER_IP_:5060 -> __OPENSIPS_IP__:5060
>>>> ACK sip:603#558533822977@_ASTERISK_IP_:5060 SIP/2.0.
>>>> Contact: .
>>>> CSeq: 1 ACK.
>>>> From: ;tag=1c512295868.
>>>>
>>>>  Message that opensips has sended
>>>>
>>>> U __OPENSIPS_IP__:5060 -> _ASTERISK_IP_:5060
>>>> ACK sip:603#558533822977@_ASTERISK_IP_:5060 SIP/2.0.
>>>> Contact: .
>>>> CSeq: 1 ACK.
>>>> From: <...S.i.~.\..)..N(..24Y3-.9.>;tag=1c512295868.
>>>>
>>>>  With this, i get on my log  messages like below.
>>>>
>>>> ERROR:core:parse_to: unexpected char [\] in status 6: <<<#032??S?i?~?>>
>>>> .
>>>> ERROR:core:parse_from_header: bad from header
>>>>  ERROR:uac:restore_uris_reply: failed to find/parse FROM hdr
>>>>
>>>>  and the acc show me this
>>>>
>>>> reason=Call leg/transaction does not exist
>>>>
>>>>  i dont understand why this is happen, but happen just some times, and
>>>> as i can find just with one customer.
>>>>
>>>>
>>>>  if somebody can point me what kind of mistake can generate this error
>>>> i will apreciate.
>>>>
>>>>  Thanks
>>>>
>>>>
>>>>  ___
>>>> Users mailing 
>>>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>>
>>>
>>>
>>
>>
>
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Re: [OpenSIPS-Users] Bad from on header

[Mike Tesliuk]

Ok Bogdan, thanks for you explanation, i will check that


2014-03-17 10:43 GMT-04:00 Bogdan-Andrei Iancu :

>  Yes, definitely that is the problem - that string is a B64 encoded
> value, so it is case sensitive. According to RFC3261, UAs must copy the RR
> params without any change (even if they do not understand). So your UAC is
> broken when comes to handling RR headers.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 17.03.2014 15:36, Mike Tesliuk wrote:
>
> Ok,
>
>  I receive the invite, and i send the invite with this vsf
> AF5CW0NKbgEAcAR4BxYLHgAABB0FGgA4
>
>  on the 180 is ok
>  on the 183 is ok
>  on the 200 is ok
>
>  The ack is ok, but lower case ( af5cw0nkbgeacar4bxylhgaabb0fgga4 )
>
>  so everything broke
>
> From: <...S.m.z._6.A..^$..4.X.-.5.>;tag=1c731545057.
>
>
>  The lower case on ACK can be the problem ?
>
>
>
> 2014-03-17 5:28 GMT-04:00 Bogdan-Andrei Iancu :
>
>>  Hello Mike,
>>
>> The restore/change of the FROM hdr is done automatically for the
>> sequential requests. What I suspect in your case is an altering of the
>> RR/Route "vsf" param - check if you have the same value in the outgoing
>> INVITE (original), in the invite 200 OK (in RR hdr) and the incoming ACK
>> (in Route hdr)
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>>   On 17.03.2014 00:03, Mike Tesliuk wrote:
>>
>>  Hello Bogdan,
>>
>>  Yes, on the initial request we have the uac_replace_from , but in this
>> case the ack is not supposed to reach the function , as i say before, this
>> does not happen on every dialog, just in some situation that i dont
>> identify exactly which one yet.
>>
>>
>> 2014-03-16 17:06 GMT-04:00 Bogdan-Andrei Iancu :
>>
>>>  Hello Mike,
>>>
>>> Are you using the uac_replace_from() for that call ?
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>
>>>  On 14.03.2014 19:48, Mike Tesliuk wrote:
>>>
>>> Hello Guys,
>>>
>>> Im checking about a problem here i hope somebody can help me.
>>>
>>>  Some times when i receive an ACK opensips is doing something strange
>>> with the from, check the mesage below
>>>
>>>  Message that opensips has received
>>>
>>> U _CUSTOMER_IP_:5060 -> __OPENSIPS_IP__:5060
>>> ACK sip:603#558533822977@_ASTERISK_IP_:5060 SIP/2.0.
>>> Contact: .
>>> CSeq: 1 ACK.
>>> From: ;tag=1c512295868.
>>>
>>>  Message that opensips has sended
>>>
>>> U __OPENSIPS_IP__:5060 -> _ASTERISK_IP_:5060
>>> ACK sip:603#558533822977@_ASTERISK_IP_:5060 SIP/2.0.
>>> Contact: .
>>> CSeq: 1 ACK.
>>> From: <...S.i.~.\..)..N(..24Y3-.9.>;tag=1c512295868.
>>>
>>>  With this, i get on my log  messages like below.
>>>
>>> ERROR:core:parse_to: unexpected char [\] in status 6: <<<#032??S?i?~?>> .
>>> ERROR:core:parse_from_header: bad from header
>>>  ERROR:uac:restore_uris_reply: failed to find/parse FROM hdr
>>>
>>>  and the acc show me this
>>>
>>> reason=Call leg/transaction does not exist
>>>
>>>  i dont understand why this is happen, but happen just some times, and
>>> as i can find just with one customer.
>>>
>>>
>>>  if somebody can point me what kind of mistake can generate this error i
>>> will apreciate.
>>>
>>>  Thanks
>>>
>>>
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>>>
>>>
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Re: [OpenSIPS-Users] Bad from on header

[Mike Tesliuk]

Looking for other dialog where the ACK does not change to lower case i have
no problems.


2014-03-17 9:36 GMT-04:00 Mike Tesliuk :

> Ok,
>
> I receive the invite, and i send the invite with this vsf
> AF5CW0NKbgEAcAR4BxYLHgAABB0FGgA4
>
> on the 180 is ok
> on the 183 is ok
> on the 200 is ok
>
> The ack is ok, but lower case ( af5cw0nkbgeacar4bxylhgaabb0fgga4 )
>
> so everything broke
>
> From: <...S.m.z._6.A..^$..4.X.-.5.>;tag=1c731545057.
>
>
> The lower case on ACK can be the problem ?
>
>
>
> 2014-03-17 5:28 GMT-04:00 Bogdan-Andrei Iancu :
>
>  Hello Mike,
>>
>> The restore/change of the FROM hdr is done automatically for the
>> sequential requests. What I suspect in your case is an altering of the
>> RR/Route "vsf" param - check if you have the same value in the outgoing
>> INVITE (original), in the invite 200 OK (in RR hdr) and the incoming ACK
>> (in Route hdr)
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 17.03.2014 00:03, Mike Tesliuk wrote:
>>
>>  Hello Bogdan,
>>
>>  Yes, on the initial request we have the uac_replace_from , but in this
>> case the ack is not supposed to reach the function , as i say before, this
>> does not happen on every dialog, just in some situation that i dont
>> identify exactly which one yet.
>>
>>
>> 2014-03-16 17:06 GMT-04:00 Bogdan-Andrei Iancu :
>>
>>>  Hello Mike,
>>>
>>> Are you using the uac_replace_from() for that call ?
>>>
>>> Regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>
>>>  On 14.03.2014 19:48, Mike Tesliuk wrote:
>>>
>>> Hello Guys,
>>>
>>> Im checking about a problem here i hope somebody can help me.
>>>
>>>  Some times when i receive an ACK opensips is doing something strange
>>> with the from, check the mesage below
>>>
>>>  Message that opensips has received
>>>
>>> U _CUSTOMER_IP_:5060 -> __OPENSIPS_IP__:5060
>>> ACK sip:603#558533822977@_ASTERISK_IP_:5060 SIP/2.0.
>>> Contact: .
>>> CSeq: 1 ACK.
>>> From: ;tag=1c512295868.
>>>
>>>  Message that opensips has sended
>>>
>>> U __OPENSIPS_IP__:5060 -> _ASTERISK_IP_:5060
>>> ACK sip:603#558533822977@_ASTERISK_IP_:5060 SIP/2.0.
>>> Contact: .
>>> CSeq: 1 ACK.
>>> From: <...S.i.~.\..)..N(..24Y3-.9.>;tag=1c512295868.
>>>
>>>  With this, i get on my log  messages like below.
>>>
>>> ERROR:core:parse_to: unexpected char [\] in status 6: <<<#032??S?i?~?>> .
>>> ERROR:core:parse_from_header: bad from header
>>>  ERROR:uac:restore_uris_reply: failed to find/parse FROM hdr
>>>
>>>  and the acc show me this
>>>
>>> reason=Call leg/transaction does not exist
>>>
>>>  i dont understand why this is happen, but happen just some times, and
>>> as i can find just with one customer.
>>>
>>>
>>>  if somebody can point me what kind of mistake can generate this error i
>>> will apreciate.
>>>
>>>  Thanks
>>>
>>>
>>>  ___
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>>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>
>>
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Re: [OpenSIPS-Users] Bad from on header

[Mike Tesliuk]

Ok,

I receive the invite, and i send the invite with this vsf
AF5CW0NKbgEAcAR4BxYLHgAABB0FGgA4

on the 180 is ok
on the 183 is ok
on the 200 is ok

The ack is ok, but lower case ( af5cw0nkbgeacar4bxylhgaabb0fgga4 )

so everything broke

From: <...S.m.z._6.A..^$..4.X.-.5.>;tag=1c731545057.


The lower case on ACK can be the problem ?



2014-03-17 5:28 GMT-04:00 Bogdan-Andrei Iancu :

>  Hello Mike,
>
> The restore/change of the FROM hdr is done automatically for the
> sequential requests. What I suspect in your case is an altering of the
> RR/Route "vsf" param - check if you have the same value in the outgoing
> INVITE (original), in the invite 200 OK (in RR hdr) and the incoming ACK
> (in Route hdr)
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 17.03.2014 00:03, Mike Tesliuk wrote:
>
>  Hello Bogdan,
>
>  Yes, on the initial request we have the uac_replace_from , but in this
> case the ack is not supposed to reach the function , as i say before, this
> does not happen on every dialog, just in some situation that i dont
> identify exactly which one yet.
>
>
> 2014-03-16 17:06 GMT-04:00 Bogdan-Andrei Iancu :
>
>>  Hello Mike,
>>
>> Are you using the uac_replace_from() for that call ?
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>>  On 14.03.2014 19:48, Mike Tesliuk wrote:
>>
>> Hello Guys,
>>
>> Im checking about a problem here i hope somebody can help me.
>>
>>  Some times when i receive an ACK opensips is doing something strange
>> with the from, check the mesage below
>>
>>  Message that opensips has received
>>
>> U _CUSTOMER_IP_:5060 -> __OPENSIPS_IP__:5060
>> ACK sip:603#558533822977@_ASTERISK_IP_:5060 SIP/2.0.
>> Contact: .
>> CSeq: 1 ACK.
>> From: ;tag=1c512295868.
>>
>>  Message that opensips has sended
>>
>> U __OPENSIPS_IP__:5060 -> _ASTERISK_IP_:5060
>> ACK sip:603#558533822977@_ASTERISK_IP_:5060 SIP/2.0.
>> Contact: .
>> CSeq: 1 ACK.
>> From: <...S.i.~.\..)..N(..24Y3-.9.>;tag=1c512295868.
>>
>>  With this, i get on my log  messages like below.
>>
>> ERROR:core:parse_to: unexpected char [\] in status 6: <<<#032??S?i?~?>> .
>> ERROR:core:parse_from_header: bad from header
>>  ERROR:uac:restore_uris_reply: failed to find/parse FROM hdr
>>
>>  and the acc show me this
>>
>> reason=Call leg/transaction does not exist
>>
>>  i dont understand why this is happen, but happen just some times, and as
>> i can find just with one customer.
>>
>>
>>  if somebody can point me what kind of mistake can generate this error i
>> will apreciate.
>>
>>  Thanks
>>
>>
>>  ___
>> Users mailing 
>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>
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Re: [OpenSIPS-Users] Bad from on header

[Mike Tesliuk]

Hello Bogdan,

Yes, on the initial request we have the uac_replace_from , but in this case
the ack is not supposed to reach the function , as i say before, this does
not happen on every dialog, just in some situation that i dont identify
exactly which one yet.


2014-03-16 17:06 GMT-04:00 Bogdan-Andrei Iancu :

>  Hello Mike,
>
> Are you using the uac_replace_from() for that call ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 14.03.2014 19:48, Mike Tesliuk wrote:
>
>Hello Guys,
>
> Im checking about a problem here i hope somebody can help me.
>
>  Some times when i receive an ACK opensips is doing something strange with
> the from, check the mesage below
>
>  Message that opensips has received
>
> U _CUSTOMER_IP_:5060 -> __OPENSIPS_IP__:5060
> ACK sip:603#558533822977@_ASTERISK_IP_:5060 SIP/2.0.
> Contact: .
> CSeq: 1 ACK.
> From: ;tag=1c512295868.
>
>  Message that opensips has sended
>
> U __OPENSIPS_IP__:5060 -> _ASTERISK_IP_:5060
> ACK sip:603#558533822977@_ASTERISK_IP_:5060 SIP/2.0.
> Contact: .
> CSeq: 1 ACK.
> From: <...S.i.~.\..)..N(..24Y3-.9.>;tag=1c512295868.
>
>  With this, i get on my log  messages like below.
>
> ERROR:core:parse_to: unexpected char [\] in status 6: <<<#032??S?i?~?>> .
> ERROR:core:parse_from_header: bad from header
>  ERROR:uac:restore_uris_reply: failed to find/parse FROM hdr
>
>  and the acc show me this
>
> reason=Call leg/transaction does not exist
>
>  i dont understand why this is happen, but happen just some times, and as
> i can find just with one customer.
>
>
>  if somebody can point me what kind of mistake can generate this error i
> will apreciate.
>
>  Thanks
>
>
> ___
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> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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Re: [OpenSIPS-Users] OpenSIPS variables

[Mike Tesliuk]

check on the documentation

http://www.opensips.org/Documentation/Script-CoreVar




2014-03-14 3:18 GMT-04:00 Mike Claudi Pedersen :

>following er some asterisk values that is formatted and set;
>
> set(FROM=${CUT(SIP_HEADER(From),:,2)});
> set(FROMIP=${CUT(FROM,@,2)});
>  set(FROM=${CUT(FROM,@,1)});
> set(TO=${CUT(SIP_HEADER(To),:,2)});
> set(TO=${CUT(TO,@,1)});
>
> noop(RDNIS: ${CALLERID(rdnis)});
> noop(NAME=${CALLERID(name)});
> set(CALLERID(name)=);
>
>
> i would like to know how to do the equivalent in openSIPS.
>
>
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[OpenSIPS-Users] Bad from on header

[Mike Tesliuk]

Hello Guys,

Im checking about a problem here i hope somebody can help me.

Some times when i receive an ACK opensips is doing something strange with
the from, check the mesage below

Message that opensips has received

U _CUSTOMER_IP_:5060 -> __OPENSIPS_IP__:5060
ACK sip:603#558533822977@_ASTERISK_IP_:5060 SIP/2.0.
Contact: .
CSeq: 1 ACK.
From: ;tag=1c512295868.

Message that opensips has sended

U __OPENSIPS_IP__:5060 -> _ASTERISK_IP_:5060
ACK sip:603#558533822977@_ASTERISK_IP_:5060 SIP/2.0.
Contact: .
CSeq: 1 ACK.
From: <...S.i.~.\..)..N(..24Y3-.9.>;tag=1c512295868.

With this, i get on my log  messages like below.

ERROR:core:parse_to: unexpected char [\] in status 6: <<<#032??S?i?~?>> .
ERROR:core:parse_from_header: bad from header
 ERROR:uac:restore_uris_reply: failed to find/parse FROM hdr

and the acc show me this

reason=Call leg/transaction does not exist

i dont understand why this is happen, but happen just some times, and as i
can find just with one customer.


if somebody can point me what kind of mistake can generate this error i
will apreciate.

Thanks
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Re: [OpenSIPS-Users] No record-route on reply's

[Mike Tesliuk]

nobody have an idea about this ?


Thanks


2014/1/6 Mike Tesliuk 

> i put some more information on syslog, this is what happens when i receive
> the bye
>
> the _EXTERNAL_IP_FS_ is the return of the $si variable  , so i have the
> $fu and $ru
>
> the gateway have the ip
> *10.255.2.21 port 5021 , and openspis have the ip 10.1.69.1:5079
> <http://10.1.69.1:5079> *
>
> Jan  6 20:15:28 gcl-ss01a /usr/sbin/opensips[31827]: We are on loose_route
> / match_dialog [ BYE ]
> Jan  6 20:15:28 gcl-ss01a /usr/sbin/opensips[31827]: informations:
> _EXTERNAL_IP_FS_ sip:200214@_EXTERNAL_IP_FS_ sip:255755813256@10.1.69.1:
> *5021*;transport=udp
> Jan  6 20:15:28 gcl-ss01a /usr/sbin/opensips[31827]: Dialog State: 5
> Jan  6 20:15:28 gcl-ss01a /usr/sbin/opensips[31827]: Dialog Reason:
> Downstream BYE
> Jan  6 20:15:28 gcl-ss01a /usr/sbin/opensips[31827]: dialog match, method
> bye ack
> Jan  6 20:15:28 gcl-ss01a /usr/sbin/opensips[31827]: Relaying a message [
> BYE ]
> Jan  6 20:15:28 gcl-ss01a /usr/sbin/opensips[31827]: Information:
> _EXTERNAL_IP_FS_ sip:200214@_EXTERNAL_IP_FS_
> sip:255755813256@10.1.69.1:5021;transport=udp
> Jan  6 20:15:29 gcl-ss01a /usr/sbin/opensips[31828]: Dialog State: 
> Jan  6 20:15:29 gcl-ss01a /usr/sbin/opensips[31828]: Dialog Reason: 
> Jan  6 20:15:29 gcl-ss01a /usr/sbin/opensips[31828]: dialog match, method
> bye ack
> Jan  6 20:15:29 gcl-ss01a /usr/sbin/opensips[31828]: Relaying a message [
> BYE ]
> Jan  6 20:15:29 gcl-ss01a /usr/sbin/opensips[31828]: Information:
> _EXTERNAL_IP_FS_ sip:200214@_EXTERNAL_IP_FS_ sip:10.1.69.1:5079
> ;did=3df.33a55a92
> Jan  6 20:15:31 gcl-ss01a /usr/sbin/opensips[31829]: Dialog State: 
> Jan  6 20:15:31 gcl-ss01a /usr/sbin/opensips[31829]: Dialog Reason: 
> Jan  6 20:15:31 gcl-ss01a /usr/sbin/opensips[31829]: dialog match, method
> bye ack
> Jan  6 20:15:31 gcl-ss01a /usr/sbin/opensips[31829]: Relaying a message [
> BYE ]
> Jan  6 20:15:31 gcl-ss01a /usr/sbin/opensips[31829]: Information:
> _EXTERNAL_IP_FS_ sip:200214@_EXTERNAL_IP_FS_ sip:10.1.69.1:5079
> ;did=3df.33a55a92
> Jan  6 20:15:40 gcl-ss01a /usr/sbin/opensips[31826]: Dialog State: 
> Jan  6 20:15:40 gcl-ss01a /usr/sbin/opensips[31826]: Dialog Reason: 
> Jan  6 20:15:40 gcl-ss01a /usr/sbin/opensips[31826]: dialog match, method
> bye ack
> Jan  6 20:15:40 gcl-ss01a /usr/sbin/opensips[31826]: Relaying a message [
> BYE ]
> Jan  6 20:15:40 gcl-ss01a /usr/sbin/opensips[31826]: Information:
> *10.255.2.21* sip:255755813256@10.1.69.1:5079sip:255755813256@10.1.69.1:5021
> ;transport=udp
>
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Re: [OpenSIPS-Users] No record-route on reply's

[Mike Tesliuk]

i put some more information on syslog, this is what happens when i receive
the bye

the _EXTERNAL_IP_FS_ is the return of the $si variable  , so i have the $fu
and $ru

the gateway have the ip
*10.255.2.21 port 5021 , and openspis have the ip 10.1.69.1:5079
*

Jan  6 20:15:28 gcl-ss01a /usr/sbin/opensips[31827]: We are on loose_route
/ match_dialog [ BYE ]
Jan  6 20:15:28 gcl-ss01a /usr/sbin/opensips[31827]: informations:
_EXTERNAL_IP_FS_ sip:200214@_EXTERNAL_IP_FS_ sip:255755813256@10.1.69.1:
*5021*;transport=udp
Jan  6 20:15:28 gcl-ss01a /usr/sbin/opensips[31827]: Dialog State: 5
Jan  6 20:15:28 gcl-ss01a /usr/sbin/opensips[31827]: Dialog Reason:
Downstream BYE
Jan  6 20:15:28 gcl-ss01a /usr/sbin/opensips[31827]: dialog match, method
bye ack
Jan  6 20:15:28 gcl-ss01a /usr/sbin/opensips[31827]: Relaying a message [
BYE ]
Jan  6 20:15:28 gcl-ss01a /usr/sbin/opensips[31827]: Information:
_EXTERNAL_IP_FS_ sip:200214@_EXTERNAL_IP_FS_ sip:255755813256@10.1.69.1:5021
;transport=udp
Jan  6 20:15:29 gcl-ss01a /usr/sbin/opensips[31828]: Dialog State: 
Jan  6 20:15:29 gcl-ss01a /usr/sbin/opensips[31828]: Dialog Reason: 
Jan  6 20:15:29 gcl-ss01a /usr/sbin/opensips[31828]: dialog match, method
bye ack
Jan  6 20:15:29 gcl-ss01a /usr/sbin/opensips[31828]: Relaying a message [
BYE ]
Jan  6 20:15:29 gcl-ss01a /usr/sbin/opensips[31828]: Information:
_EXTERNAL_IP_FS_ sip:200214@_EXTERNAL_IP_FS_ sip:10.1.69.1:5079
;did=3df.33a55a92
Jan  6 20:15:31 gcl-ss01a /usr/sbin/opensips[31829]: Dialog State: 
Jan  6 20:15:31 gcl-ss01a /usr/sbin/opensips[31829]: Dialog Reason: 
Jan  6 20:15:31 gcl-ss01a /usr/sbin/opensips[31829]: dialog match, method
bye ack
Jan  6 20:15:31 gcl-ss01a /usr/sbin/opensips[31829]: Relaying a message [
BYE ]
Jan  6 20:15:31 gcl-ss01a /usr/sbin/opensips[31829]: Information:
_EXTERNAL_IP_FS_ sip:200214@_EXTERNAL_IP_FS_ sip:10.1.69.1:5079
;did=3df.33a55a92
Jan  6 20:15:40 gcl-ss01a /usr/sbin/opensips[31826]: Dialog State: 
Jan  6 20:15:40 gcl-ss01a /usr/sbin/opensips[31826]: Dialog Reason: 
Jan  6 20:15:40 gcl-ss01a /usr/sbin/opensips[31826]: dialog match, method
bye ack
Jan  6 20:15:40 gcl-ss01a /usr/sbin/opensips[31826]: Relaying a message [
BYE ]
Jan  6 20:15:40 gcl-ss01a /usr/sbin/opensips[31826]: Information:
*10.255.2.21* sip:255755813256@10.1.69.1:5079sip:255755813256@10.1.69.1:5021
;transport=udp
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Re: [OpenSIPS-Users] No record-route on reply's

[Mike Tesliuk]

Ok, the dialog are ok using topology_hiding

The only problem that i see now is with the bye.

This is what i have on my syslog

Jan  6 17:39:29 gcl-ss01a /usr/sbin/opensips[29909]: Initial Invite
sip:200214@PUBLIC_IP sip:255755813256@10.1.69.1:5079 [ INVITE ]
Jan  6 17:39:29 gcl-ss01a /usr/sbin/opensips[29909]: Executing
record_route()
Jan  6 17:39:29 gcl-ss01a /usr/sbin/opensips[29909]: IP PUBLIC_IP
authorized [ INVITE ]
Jan  6 17:39:29 gcl-ss01a /usr/sbin/opensips[29909]: GROUPID TO
255755813256 IS 25575
Jan  6 17:39:29 gcl-ss01a /usr/sbin/opensips[29909]: Call to:
sip:255755813256@10.1.69.1:5079 [ INVITE ]
Jan  6 17:39:29 gcl-ss01a /usr/sbin/opensips[29909]: IP Source: PUBLIC_IP [
INVITE ]
Jan  6 17:39:29 gcl-ss01a /usr/sbin/opensips[29909]: IP Destination:
10.1.69.1 [ INVITE ]
Jan  6 17:39:29 gcl-ss01a /usr/sbin/opensips[29909]: Call to 255755813256 [
INVITE ]
Jan  6 17:39:29 gcl-ss01a /usr/sbin/opensips[29909]: route 3
Jan  6 17:39:29 gcl-ss01a /usr/sbin/opensips[29909]: Preparing call to sip:
10.255.2.21:5021
Jan  6 17:39:29 gcl-ss01a /usr/sbin/opensips[29909]: Branch
sip:255755813256@10.1.69.1:5079 [ INVITE ]
Jan  6 17:39:29 gcl-ss01a /usr/sbin/opensips[29910]: Reply received from
10.255.2.21 - sip:200214@PUBLIC_IP   with method INVITE
Jan  6 17:39:31 gcl-ss01a /usr/sbin/opensips[29911]: Reply received from
10.255.2.21 - sip:200214@PUBLIC_IP   with method INVITE
Jan  6 17:39:41 gcl-ss01a /usr/sbin/opensips[29912]: Reply received from
10.255.2.21 - sip:200214@PUBLIC_IP   with method INVITE
Jan  6 17:39:41 gcl-ss01a /usr/sbin/opensips[29909]: We are on loose_route
/ match_dialog [ ACK ]
Jan  6 17:39:41 gcl-ss01a /usr/sbin/opensips[29909]: Dialog State: 4
Jan  6 17:39:41 gcl-ss01a /usr/sbin/opensips[29909]: Dialog Reason: 
Jan  6 17:39:41 gcl-ss01a /usr/sbin/opensips[29909]: dialog match, method
bye ack
Jan  6 17:39:41 gcl-ss01a /usr/sbin/opensips[29909]: Relaying a message [
ACK ]
Jan  6 17:39:46 gcl-ss01a /usr/sbin/opensips[29910]: We are on loose_route
/ match_dialog [ BYE ]
Jan  6 17:39:46 gcl-ss01a /usr/sbin/opensips[29910]: Dialog State: 5
Jan  6 17:39:46 gcl-ss01a /usr/sbin/opensips[29910]: Dialog Reason:
Downstream BYE

here, the dialog is destroyed , but the bye dont go to the gateway

Jan  6 17:39:46 gcl-ss01a /usr/sbin/opensips[29910]: dialog match, method
bye ack
Jan  6 17:39:46 gcl-ss01a /usr/sbin/opensips[29910]: Relaying a message [
BYE ]
Jan  6 17:39:47 gcl-ss01a /usr/sbin/opensips[29911]: Relaying a message [
BYE ]
Jan  6 17:39:49 gcl-ss01a /usr/sbin/opensips[29912]: Relaying a message [
BYE ]
Jan  6 17:39:51 gcl-ss01a /usr/sbin/opensips[29909]: Relaying a message [
BYE ]



using ngrep i see a bye coming from the gateway, but no the opensips
sending the bye

i have no ideas anymore about this, if somebody could help i will apreciate
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Re: [OpenSIPS-Users] No record-route on reply's

[Mike Tesliuk]

ok, i try to use this and i got diferent results now

topology_hiding("U");


i didnt see the dialog errors anymore , i got a problem with the bye now
but i will made some changes on the network to check about this.

i will post the result later




2014/1/6 Mike Tesliuk 

> As i receive the 200 ok without the Record-Route by the gateway, is it
> possible to the gateway stablish the signalling directly with the user and
> in this case i didnt receive the bye ?
>
> i think that is what happen with this gateway
>
>
> 2014/1/5 Mike Tesliuk 
>
>> Hello Razvan (and everybody),
>>
>> I try this, the dialog seems to be ok because the dialog is beeing
>> deleted, but i got this messages on syslog
>>
>> Jan  5 22:54:55 gcl-ss01a /usr/sbin/opensips[23924]:
>> ERROR:rr:get_remote_target: Invalid routing type - 0
>> Jan  5 22:54:55 gcl-ss01a /usr/sbin/opensips[23924]:
>> ERROR:dialog:dlg_validate_dialog: failed fetching remote target from msg
>> Jan  5 22:54:55 gcl-ss01a /usr/sbin/opensips[23924]: In-Dialog BYE from
>> XX.XX.XX.XX (callid=1b1c3705-f0ff-1231-c396-001a4bd5a0b4) is not valid
>> according to dialog
>>
>>
>>
>> in this case, the user send the bye
>>
>> U __IP__CUSTOMER__:30664 -> __IP__OPENSIPS__:5069
>> BYE sip:255755813256@__IP__OPENSIPS__:5069;transport=udp SIP/2.0.
>> Via: SIP/2.0/UDP
>> __IP__CUSTOMER__:30664;branch=z9hG4bK-d8754z-2d4c2058767ff73a-1---d8754z-;rport.
>> Max-Forwards: 70.
>> Contact: .
>> To: ;tag=ZSag78XKZQ6yD.
>> From: ;tag=ea263e5b.
>> Call-ID: OGFiYzliMGIzZmYwOTczY2E2YTg1OWYwNzZhMDVlMTA..
>> CSeq: 3 BYE.
>> Proxy-Authorization: Digest
>> username="200214",realm="__IP__OPENSIPS__",nonce="8ef536ce-d39b-4036-90d1-04c54fe9e133",uri="sip:255755813256@
>> __IP__OPENSIPS__:5069;transport=udp",response="72ef0971dda591f6b7684b375a5a3ac1",cnonce="8db354c7dc934c13b3577ff5db18297c",nc=0002,qop=auth,algorithm=MD5.
>> User-Agent: Bria Professional release 2.4 stamp 49381.
>> Reason: SIP;description="User Hung Up".
>> Content-Length: 0.
>>
>>
>> And on opensips this is what i have.
>>
>> #
>> U __IP__GATEWAY__:39040 -> __IP_OPENSIPS:5079
>> BYE sip:255755813256@__IP_OPENSIPS:5021;transport=udp SIP/2.0.
>> Via: SIP/2.0/UDP 10.255.2.21:5021
>> ;branch=z9hG4bKervg1296366635;received=10.255.2.21.
>> From: ;tag=12ab34cd.
>> To: "200214" ;tag=023883eQv0vHS.
>> Call-ID: 1b1c3705-f0ff-1231-c396-001a4bd5a0b4.
>> CSeq: 23 BYE.
>> Max-Forwards: 70.
>> Content-Length: 0.
>> .
>>
>> #
>> U __IP_OPENSIPS:5079 -> __IP__GATEWAY__:39040
>> SIP/2.0 481 Call Does Not Exist.
>> Via: SIP/2.0/UDP 10.255.2.21:5021
>> ;branch=z9hG4bKervg1296366635;rport=39040;received=__IP__GATEWAY__.
>> From: ;tag=12ab34cd.
>> To: "200214" ;tag=023883eQv0vHS.
>> Call-ID: 1b1c3705-f0ff-1231-c396-001a4bd5a0b4.
>> CSeq: 23 BYE.
>>
>> User-Agent: vBilling.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, NOTIFY.
>> Supported: precondition, path, replaces.
>>  Content-Length: 0.
>>
>>
>>
>> The call-ID on sip is ok, it is the same of the invite, session progress
>> etc..
>>
>>
>> if you guys have any tip I will apreciate, this is a new situation for
>> me, happen just with this gateway (I dont remember the brand now , it is a
>> friend enviroment and Im trying to help)
>>
>>
>> 2013/12/25 Mike Tesliuk 
>>
>>> Hello Razvan,
>>>
>>> thank you for your help, i check about this function before, i will try
>>> that and i let you know if solve , thank you and happy hollidays
>>>
>>>
>>> 2013/12/24 Răzvan Crainea 
>>>
>>>> Hi, Mike!
>>>>
>>>> Have you tried matching the dialogs using the match_dialog()
>>>> function[1]? Also, for sequential requests, you should try using the
>>>> fix_route_dialog() function[2].
>>>>
>>>> [1] http://www.opensips.org/html/docs/modules/1.10.x/dialog.
>>>> html#id295144
>>>> [2] http://www.opensips.org/html/docs/modules/1.10.x/dialog.
>>>> html#id295287
>>>>
>>>> Best regards,
>>>>
>>>> Răzvan Crainea
>>>> OpenSIPS Core Developer
>>>> http://www.opensips-solutions.com
>>>>
>>>>
>>>> On 12/21/2013 06:25 PM, Mike Tesliuk wrote:
>>>>
>>>>

Re: [OpenSIPS-Users] No record-route on reply's

[Mike Tesliuk]

As i receive the 200 ok without the Record-Route by the gateway, is it
possible to the gateway stablish the signalling directly with the user and
in this case i didnt receive the bye ?

i think that is what happen with this gateway


2014/1/5 Mike Tesliuk 

> Hello Razvan (and everybody),
>
> I try this, the dialog seems to be ok because the dialog is beeing
> deleted, but i got this messages on syslog
>
> Jan  5 22:54:55 gcl-ss01a /usr/sbin/opensips[23924]:
> ERROR:rr:get_remote_target: Invalid routing type - 0
> Jan  5 22:54:55 gcl-ss01a /usr/sbin/opensips[23924]:
> ERROR:dialog:dlg_validate_dialog: failed fetching remote target from msg
> Jan  5 22:54:55 gcl-ss01a /usr/sbin/opensips[23924]: In-Dialog BYE from
> XX.XX.XX.XX (callid=1b1c3705-f0ff-1231-c396-001a4bd5a0b4) is not valid
> according to dialog
>
>
>
> in this case, the user send the bye
>
> U __IP__CUSTOMER__:30664 -> __IP__OPENSIPS__:5069
> BYE sip:255755813256@__IP__OPENSIPS__:5069;transport=udp SIP/2.0.
> Via: SIP/2.0/UDP
> __IP__CUSTOMER__:30664;branch=z9hG4bK-d8754z-2d4c2058767ff73a-1---d8754z-;rport.
> Max-Forwards: 70.
> Contact: .
> To: ;tag=ZSag78XKZQ6yD.
> From: ;tag=ea263e5b.
> Call-ID: OGFiYzliMGIzZmYwOTczY2E2YTg1OWYwNzZhMDVlMTA..
> CSeq: 3 BYE.
> Proxy-Authorization: Digest
> username="200214",realm="__IP__OPENSIPS__",nonce="8ef536ce-d39b-4036-90d1-04c54fe9e133",uri="sip:255755813256@
> __IP__OPENSIPS__:5069;transport=udp",response="72ef0971dda591f6b7684b375a5a3ac1",cnonce="8db354c7dc934c13b3577ff5db18297c",nc=0002,qop=auth,algorithm=MD5.
> User-Agent: Bria Professional release 2.4 stamp 49381.
> Reason: SIP;description="User Hung Up".
> Content-Length: 0.
>
>
> And on opensips this is what i have.
>
> #
> U __IP__GATEWAY__:39040 -> __IP_OPENSIPS:5079
> BYE sip:255755813256@__IP_OPENSIPS:5021;transport=udp SIP/2.0.
> Via: SIP/2.0/UDP 10.255.2.21:5021
> ;branch=z9hG4bKervg1296366635;received=10.255.2.21.
> From: ;tag=12ab34cd.
> To: "200214" ;tag=023883eQv0vHS.
> Call-ID: 1b1c3705-f0ff-1231-c396-001a4bd5a0b4.
> CSeq: 23 BYE.
> Max-Forwards: 70.
> Content-Length: 0.
> .
>
> #
> U __IP_OPENSIPS:5079 -> __IP__GATEWAY__:39040
> SIP/2.0 481 Call Does Not Exist.
> Via: SIP/2.0/UDP 10.255.2.21:5021
> ;branch=z9hG4bKervg1296366635;rport=39040;received=__IP__GATEWAY__.
> From: ;tag=12ab34cd.
> To: "200214" ;tag=023883eQv0vHS.
> Call-ID: 1b1c3705-f0ff-1231-c396-001a4bd5a0b4.
> CSeq: 23 BYE.
>
> User-Agent: vBilling.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
> NOTIFY.
> Supported: precondition, path, replaces.
> Content-Length: 0.
>
>
>
> The call-ID on sip is ok, it is the same of the invite, session progress
> etc..
>
>
> if you guys have any tip I will apreciate, this is a new situation for me,
> happen just with this gateway (I dont remember the brand now , it is a
> friend enviroment and Im trying to help)
>
>
> 2013/12/25 Mike Tesliuk 
>
>> Hello Razvan,
>>
>> thank you for your help, i check about this function before, i will try
>> that and i let you know if solve , thank you and happy hollidays
>>
>>
>> 2013/12/24 Răzvan Crainea 
>>
>>> Hi, Mike!
>>>
>>> Have you tried matching the dialogs using the match_dialog()
>>> function[1]? Also, for sequential requests, you should try using the
>>> fix_route_dialog() function[2].
>>>
>>> [1] http://www.opensips.org/html/docs/modules/1.10.x/dialog.
>>> html#id295144
>>> [2] http://www.opensips.org/html/docs/modules/1.10.x/dialog.
>>> html#id295287
>>>
>>> Best regards,
>>>
>>> Răzvan Crainea
>>> OpenSIPS Core Developer
>>> http://www.opensips-solutions.com
>>>
>>>
>>> On 12/21/2013 06:25 PM, Mike Tesliuk wrote:
>>>
>>>> Hello Guys,
>>>>
>>>>
>>>> Im getting a strange situation here that i dont know how to deal
>>>>
>>>> i have an enviroment where i have freeswitch receiving a call to billing
>>>> and opensips doing the load_balance to the gateways.
>>>>
>>>> When i send the call to the gateway i receive the reply without the
>>>> record-route header, i try to put  an asterisk server as gateway and
>>>> this not happen in this scenario .
>>>>
>>>> Below the invite that i send to the gateway
>>>>
>>>> U 10.1.69.1:5079 <http://10.1.69.1:5079> -> 10.255.2.31:5031
>>>> <http://10.255.2.31:5

Re: [OpenSIPS-Users] No record-route on reply's

[Mike Tesliuk]

Hello Razvan (and everybody),

I try this, the dialog seems to be ok because the dialog is beeing deleted,
but i got this messages on syslog

Jan  5 22:54:55 gcl-ss01a /usr/sbin/opensips[23924]:
ERROR:rr:get_remote_target: Invalid routing type - 0
Jan  5 22:54:55 gcl-ss01a /usr/sbin/opensips[23924]:
ERROR:dialog:dlg_validate_dialog: failed fetching remote target from msg
Jan  5 22:54:55 gcl-ss01a /usr/sbin/opensips[23924]: In-Dialog BYE from
XX.XX.XX.XX (callid=1b1c3705-f0ff-1231-c396-001a4bd5a0b4) is not valid
according to dialog



in this case, the user send the bye

U __IP__CUSTOMER__:30664 -> __IP__OPENSIPS__:5069
BYE sip:255755813256@__IP__OPENSIPS__:5069;transport=udp SIP/2.0.
Via: SIP/2.0/UDP
__IP__CUSTOMER__:30664;branch=z9hG4bK-d8754z-2d4c2058767ff73a-1---d8754z-;rport.
Max-Forwards: 70.
Contact: .
To: ;tag=ZSag78XKZQ6yD.
From: ;tag=ea263e5b.
Call-ID: OGFiYzliMGIzZmYwOTczY2E2YTg1OWYwNzZhMDVlMTA..
CSeq: 3 BYE.
Proxy-Authorization: Digest
username="200214",realm="__IP__OPENSIPS__",nonce="8ef536ce-d39b-4036-90d1-04c54fe9e133",uri="sip:255755813256@
__IP__OPENSIPS__:5069;transport=udp",response="72ef0971dda591f6b7684b375a5a3ac1",cnonce="8db354c7dc934c13b3577ff5db18297c",nc=0002,qop=auth,algorithm=MD5.
User-Agent: Bria Professional release 2.4 stamp 49381.
Reason: SIP;description="User Hung Up".
Content-Length: 0.


And on opensips this is what i have.

#
U __IP__GATEWAY__:39040 -> __IP_OPENSIPS:5079
BYE sip:255755813256@__IP_OPENSIPS:5021;transport=udp SIP/2.0.
Via: SIP/2.0/UDP 10.255.2.21:5021
;branch=z9hG4bKervg1296366635;received=10.255.2.21.
From: ;tag=12ab34cd.
To: "200214" ;tag=023883eQv0vHS.
Call-ID: 1b1c3705-f0ff-1231-c396-001a4bd5a0b4.
CSeq: 23 BYE.
Max-Forwards: 70.
Content-Length: 0.
.

#
U __IP_OPENSIPS:5079 -> __IP__GATEWAY__:39040
SIP/2.0 481 Call Does Not Exist.
Via: SIP/2.0/UDP 10.255.2.21:5021
;branch=z9hG4bKervg1296366635;rport=39040;received=__IP__GATEWAY__.
From: ;tag=12ab34cd.
To: "200214" ;tag=023883eQv0vHS.
Call-ID: 1b1c3705-f0ff-1231-c396-001a4bd5a0b4.
CSeq: 23 BYE.
User-Agent: vBilling.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
NOTIFY.
Supported: precondition, path, replaces.
Content-Length: 0.



The call-ID on sip is ok, it is the same of the invite, session progress
etc..


if you guys have any tip I will apreciate, this is a new situation for me,
happen just with this gateway (I dont remember the brand now , it is a
friend enviroment and Im trying to help)


2013/12/25 Mike Tesliuk 

> Hello Razvan,
>
> thank you for your help, i check about this function before, i will try
> that and i let you know if solve , thank you and happy hollidays
>
>
> 2013/12/24 Răzvan Crainea 
>
>> Hi, Mike!
>>
>> Have you tried matching the dialogs using the match_dialog() function[1]?
>> Also, for sequential requests, you should try using the fix_route_dialog()
>> function[2].
>>
>> [1] http://www.opensips.org/html/docs/modules/1.10.x/dialog.html#id295144
>> [2] http://www.opensips.org/html/docs/modules/1.10.x/dialog.html#id295287
>>
>> Best regards,
>>
>> Răzvan Crainea
>> OpenSIPS Core Developer
>> http://www.opensips-solutions.com
>>
>>
>> On 12/21/2013 06:25 PM, Mike Tesliuk wrote:
>>
>>> Hello Guys,
>>>
>>>
>>> Im getting a strange situation here that i dont know how to deal
>>>
>>> i have an enviroment where i have freeswitch receiving a call to billing
>>> and opensips doing the load_balance to the gateways.
>>>
>>> When i send the call to the gateway i receive the reply without the
>>> record-route header, i try to put  an asterisk server as gateway and
>>> this not happen in this scenario .
>>>
>>> Below the invite that i send to the gateway
>>>
>>> U 10.1.69.1:5079 <http://10.1.69.1:5079> -> 10.255.2.31:5031
>>> <http://10.255.2.31:5031>
>>> INVITE sip:255755813256@10.1.69.1:5079
>>> <http://sip:255755813256@10.1.69.1:5079> SIP/2.0.
>>>
>>> Record-Route: >> 723c6252>.
>>> Via: SIP/2.0/UDP 10.1.69.1:5079;branch=z9hG4bKe98.72455346.0.
>>> Via: SIP/2.0/UDP
>>> 10.1.69.1:5069;received=10.1.69.1;rport=5069;branch=
>>> z9hG4bKK5N8yU10cgage.
>>> Max-Forwards: 68.
>>> From: "200214" >> <mailto:sip%3A200214@10.1.69.1>>;tag=HgcSt10Xa854e.
>>> To: >> <http://sip:255755813256@10.1.69.1:5079>>.
>>>
>>> Call-ID: 4c6591da-e483-1231-6cb4-001a4bd5a0b4.
>>> CSeq: 53458861 INVITE.
>>> Contact: .
>>> User-Agent: vBilling.
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTION

Re: [OpenSIPS-Users] No record-route on reply's

[Mike Tesliuk]

Hello Razvan,

thank you for your help, i check about this function before, i will try
that and i let you know if solve , thank you and happy hollidays


2013/12/24 Răzvan Crainea 

> Hi, Mike!
>
> Have you tried matching the dialogs using the match_dialog() function[1]?
> Also, for sequential requests, you should try using the fix_route_dialog()
> function[2].
>
> [1] http://www.opensips.org/html/docs/modules/1.10.x/dialog.html#id295144
> [2] http://www.opensips.org/html/docs/modules/1.10.x/dialog.html#id295287
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
>
> On 12/21/2013 06:25 PM, Mike Tesliuk wrote:
>
>> Hello Guys,
>>
>>
>> Im getting a strange situation here that i dont know how to deal
>>
>> i have an enviroment where i have freeswitch receiving a call to billing
>> and opensips doing the load_balance to the gateways.
>>
>> When i send the call to the gateway i receive the reply without the
>> record-route header, i try to put  an asterisk server as gateway and
>> this not happen in this scenario .
>>
>> Below the invite that i send to the gateway
>>
>> U 10.1.69.1:5079 <http://10.1.69.1:5079> -> 10.255.2.31:5031
>> <http://10.255.2.31:5031>
>> INVITE sip:255755813256@10.1.69.1:5079
>> <http://sip:255755813256@10.1.69.1:5079> SIP/2.0.
>>
>> Record-Route: > 723c6252>.
>> Via: SIP/2.0/UDP 10.1.69.1:5079;branch=z9hG4bKe98.72455346.0.
>> Via: SIP/2.0/UDP
>> 10.1.69.1:5069;received=10.1.69.1;rport=5069;branch=z9hG4bKK5N8yU10cgage.
>> Max-Forwards: 68.
>> From: "200214" > <mailto:sip%3A200214@10.1.69.1>>;tag=HgcSt10Xa854e.
>> To: > <http://sip:255755813256@10.1.69.1:5079>>.
>>
>> Call-ID: 4c6591da-e483-1231-6cb4-001a4bd5a0b4.
>> CSeq: 53458861 INVITE.
>> Contact: .
>> User-Agent: vBilling.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
>> REGISTER, NOTIFY.
>> Supported: precondition, path, replaces.
>> Allow-Events: talk, hold, conference, refer.
>> Content-Type: application/sdp.
>> Content-Disposition: session.
>> Content-Length: 195.
>> X-FS-Support: update_display,send_info.
>> Remote-Party-ID: "200214" > <mailto:sip%3A200214@10.1.69.1>>;party=calling;screen=yes;privacy=off.
>>
>>
>>
>> and below the 200 ok that i receive
>>
>> U 10.255.2.31:5031 <http://10.255.2.31:5031> -> 10.1.69.1:5079
>> <http://10.1.69.1:5079>
>>
>> SIP/2.0 200 OK.
>> Via: SIP/2.0/UDP
>> 10.1.69.1:5079;branch=z9hG4bKe98.72455346.0;received=10.1.69.1.
>> Via: SIP/2.0/UDP
>> 10.1.69.1:5069;received=10.1.69.1;rport=5069;branch=z9hG4bKK5N8yU10cgage.
>> To: > <http://sip:255755813256@10.1.69.1:5079>>;tag=12ab34cd.
>> From: "200214" > <mailto:sip%3A200214@10.1.69.1>>;tag=HgcSt10Xa854e.
>>
>> CSeq: 53458861 INVITE.
>> Call-ID: 4c6591da-e483-1231-6cb4-001a4bd5a0b4.
>> Allow: INVITE, BYE, CANCEL, ACK, INFO, REGISTER.
>> Supported:.
>> Allow-Events: telephone-event.
>> Contact: .
>> Content-Type: application/sdp.
>> Content-Length: 196.
>>
>> when i send the call to this gateway the loose route did not execute,
>> and i get error's on dialog because the dialog is not matched
>>
>>
>> how should i deal with a situation like this ?
>>
>>
>>
>>
>>
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
___
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[OpenSIPS-Users] No record-route on reply's

[Mike Tesliuk]

Hello Guys,


Im getting a strange situation here that i dont know how to deal

i have an enviroment where i have freeswitch receiving a call to billing
and opensips doing the load_balance to the gateways.

When i send the call to the gateway i receive the reply without the
record-route header, i try to put  an asterisk server as gateway and this
not happen in this scenario .

Below the invite that i send to the gateway

U 10.1.69.1:5079 -> 10.255.2.31:5031
INVITE sip:255755813256@10.1.69.1:5079 SIP/2.0.
Record-Route: .
Via: SIP/2.0/UDP 10.1.69.1:5079;branch=z9hG4bKe98.72455346.0.
Via: SIP/2.0/UDP 10.1.69.1:5069
;received=10.1.69.1;rport=5069;branch=z9hG4bKK5N8yU10cgage.
Max-Forwards: 68.
From: "200214" ;tag=HgcSt10Xa854e.
To: .
Call-ID: 4c6591da-e483-1231-6cb4-001a4bd5a0b4.
CSeq: 53458861 INVITE.
Contact: .
User-Agent: vBilling.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
NOTIFY.
Supported: precondition, path, replaces.
Allow-Events: talk, hold, conference, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 195.
X-FS-Support: update_display,send_info.
Remote-Party-ID: "200214" ;party=calling;screen=yes;privacy=off.


and below the 200 ok that i receive

U 10.255.2.31:5031 -> 10.1.69.1:5079
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 10.1.69.1:5079
;branch=z9hG4bKe98.72455346.0;received=10.1.69.1.
Via: SIP/2.0/UDP 10.1.69.1:5069
;received=10.1.69.1;rport=5069;branch=z9hG4bKK5N8yU10cgage.
To: ;tag=12ab34cd.
From: "200214" ;tag=HgcSt10Xa854e.
CSeq: 53458861 INVITE.
Call-ID: 4c6591da-e483-1231-6cb4-001a4bd5a0b4.
Allow: INVITE, BYE, CANCEL, ACK, INFO, REGISTER.
Supported:.
Allow-Events: telephone-event.
Contact: .
Content-Type: application/sdp.
Content-Length: 196.

when i send the call to this gateway the loose route did not execute, and i
get error's on dialog because the dialog is not matched


how should i deal with a situation like this ?
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Re: [OpenSIPS-Users] question about cachedb_sql

[Mike Tesliuk]

oh! , just sql:// ? i was trying with sql://localhost/my_database :)

i will try that, thanks bogdan


2013/12/10 Bogdan-Andrei Iancu 

>  Hi Mike,
>
> You need first to configure the cachedb_sql, to tell it where the SQL db
> is:
> modparam("cachedb_sql", "db_url","mysql://localhost/my_database");
>
> Then, in the dialog module, you need to point the noSQL connector to the
> previous module:
> modparam("dialog", "cachedb_url", "sql://")
>
> Give it a try and let me know.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 07.12.2013 17:03, Mike Tesliuk wrote:
>
>   Hello Guys,
>
>  I try to implement (for test purpose) the cachedb_sql but i cant use this
> on dialog module, i start the module , i have the database , but when i
> call over dialog i put the cachedb_url as "sql://localhost/opensips" (im
> using sql because is what beeing exported on debug )
>
> Dec  5 23:41:47 [5953] DBG:core:register_cachedb: registered cachedb
> system [sql]
> Dec  5 23:41:47 [5953] DBG:core:parse_cachedb_url: parsing [sql://]
> Dec  5 23:41:47 [5953] DBG:core:parse_cachedb_url: Just scheme, no actual
> url
>
>  so, when i try to call this on dialog i receive
>
> Dec  5 23:41:47 [5953] ERROR:cachedb_sql:dbcache_new_connection: bogus url
> for local cachedb
> Dec  5 23:41:47 [5953] ERROR:core:cachedb_do_init: failed to open
> connection
> Dec  5 23:41:47 [5953] ERROR:dialog:init_cachedb_utils: cannot connect to
> cachedb_url sql://localhost/cachedb
> Dec  5 23:41:47 [5953] ERROR:dialog:mod_init: cannot init cachedb utils
> Dec  5 23:41:47 [5953] ERROR:core:init_mod: failed to initialize module
> dialog
> Dec  5 23:41:47 [5953] ERROR:core:main: error while initializing modules
>
>
>
>  so, there is not possible to use cachedb_sql with dialog or im doing
> something wrong ?
>
>
>
>
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>
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Re: [OpenSIPS-Users] loadbalance share stats

[Mike Tesliuk]

Work very well with the redis, but about the dialog replication, on real
server the process is tooking like seconds to send the dialog to the other
side, there is some limitation of using the binary replication through the
web ?

Thanks.


2013/12/5 Mike Tesliuk 

> Hum, very interesting Bogdan, i will try this,
>
> Thaks for you reply.
>
>
> 2013/12/5 Bogdan-Andrei Iancu 
>
>>  Hello Mike,
>>
>> Instead of replicated the whole dialogs, you can only share (between the
>> servers) the dialog profiles which are used by the LB module. How to do
>> this?
>> 1) in dialog module, set the cachedb url pointing to a noSQL db (both
>> opensips pointing to the same)
>> 2) in LB definition, put "/s" at the end of the resource names.
>>
>> Best Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 05.12.2013 06:10, Mike Tesliuk wrote:
>>
>>  I try with the db_mode 1 on dialog and replication trought bin
>> interface and work as i expect.
>>
>>  Thanks
>>
>>
>> 2013/12/4 Mike Tesliuk 
>>
>>>  Hello Guys,
>>>
>>>  There is a way to share the loadbalance information between diferent
>>> servers that are using the same gateways ?
>>>
>>>
>>>
>>
>>
>> ___
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>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>
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[OpenSIPS-Users] question about cachedb_sql

[Mike Tesliuk]

Hello Guys,

I try to implement (for test purpose) the cachedb_sql but i cant use this
on dialog module, i start the module , i have the database , but when i
call over dialog i put the cachedb_url as "sql://localhost/opensips" (im
using sql because is what beeing exported on debug )

Dec  5 23:41:47 [5953] DBG:core:register_cachedb: registered cachedb system
[sql]
Dec  5 23:41:47 [5953] DBG:core:parse_cachedb_url: parsing [sql://]
Dec  5 23:41:47 [5953] DBG:core:parse_cachedb_url: Just scheme, no actual
url

so, when i try to call this on dialog i receive

Dec  5 23:41:47 [5953] ERROR:cachedb_sql:dbcache_new_connection: bogus url
for local cachedb
Dec  5 23:41:47 [5953] ERROR:core:cachedb_do_init: failed to open connection
Dec  5 23:41:47 [5953] ERROR:dialog:init_cachedb_utils: cannot connect to
cachedb_url sql://localhost/cachedb
Dec  5 23:41:47 [5953] ERROR:dialog:mod_init: cannot init cachedb utils
Dec  5 23:41:47 [5953] ERROR:core:init_mod: failed to initialize module
dialog
Dec  5 23:41:47 [5953] ERROR:core:main: error while initializing modules



so, there is not possible to use cachedb_sql with dialog or im doing
something wrong ?
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Re: [OpenSIPS-Users] Unable to get presence and offline messages support work in my opensips.

[Mike Tesliuk]

did you try the m_dump with $fu ?




2013/12/6 Kashif Ali Siddiqui 

> Hello Guys,
>
> I am working on a opensips deployment with DB authentication, presence,
> and offline messaging. I am currently using ver 1.9.x.
>
> Can anyone look into my conf script (attached), and point out why the
> presence and offline messaging is not working. Also let me know the
> corrections.
>
> Thanks
>
> ---
> Kashif Ali Siddiqui
> ---
>
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Re: [OpenSIPS-Users] loadbalance share stats

[Mike Tesliuk]

Hum, very interesting Bogdan, i will try this,

Thaks for you reply.


2013/12/5 Bogdan-Andrei Iancu 

>  Hello Mike,
>
> Instead of replicated the whole dialogs, you can only share (between the
> servers) the dialog profiles which are used by the LB module. How to do
> this?
> 1) in dialog module, set the cachedb url pointing to a noSQL db (both
> opensips pointing to the same)
> 2) in LB definition, put "/s" at the end of the resource names.
>
> Best Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 05.12.2013 06:10, Mike Tesliuk wrote:
>
>  I try with the db_mode 1 on dialog and replication trought bin interface
> and work as i expect.
>
>  Thanks
>
>
> 2013/12/4 Mike Tesliuk 
>
>>  Hello Guys,
>>
>>  There is a way to share the loadbalance information between diferent
>> servers that are using the same gateways ?
>>
>>
>>
>
>
> ___
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>
>
>
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Re: [OpenSIPS-Users] loadbalance share stats

[Mike Tesliuk]

I try with the db_mode 1 on dialog and replication trought bin interface
and work as i expect.

Thanks


2013/12/4 Mike Tesliuk 

> Hello Guys,
>
> There is a way to share the loadbalance information between diferent
> servers that are using the same gateways ?
>
>
>
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[OpenSIPS-Users] loadbalance share stats

[Mike Tesliuk]

Hello Guys,

There is a way to share the loadbalance information between diferent
servers that are using the same gateways ?
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Re: [OpenSIPS-Users] tests with binary replication

[Mike Tesliuk]

2013/12/3 Liviu Chircu 

>  Hello Mike,
>


Hello Liviu,

Thanks for you answer.



>
> It is very improbable that either received socket (for caller and callee)
> is NULL. A more probable error is that your 2nd OpenSIPS instance is not
> listening on the same interface(in other words, a virtual IP), the sockets
> don't match, and the packet is discarded.
>
>
yes, it is exactly whats happening




> As a side note, replication will not work at all without a common virtual
> IP belonging to both instances at the same time. Maybe the documentation
> needs some improvements...
>

yeah, this was not clear to me, but now i understand


>
> What worries me more is that this actually causes a crash! What do the
> OpenSIPS logs say? We must figure out which process crashed first.
>

The version of opensips used was the 1.11 from deb repository (repository
1.10 is giving the version 1.11), so i will try with 1.10 from source and i
can tell if this happens


>
> PS: I starred the email, but forgot to answer...
>

hehe, thank you.



>
> Best regards,
>
> --
> Liviu Chircu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 03.12.2013 17:37, Mike Tesliuk wrote:
>
> So, nobody have an idea about this ?
>
>
> 2013/12/1 Mike Tesliuk 
>
>>  Hello Guys,
>>
>> Im trying to use the binary interface to test, but when the replication
>> happens i get this on log file.
>>
>> Dec  1 22:20:58 gcl-ss01a /usr/sbin/opensips[12322]:
>> ERROR:dialog:dlg_replicated_create: Dialog in DB doesn't match any
>> listening sockets
>> Dec  1 22:20:58 gcl-ss01a /usr/sbin/opensips[12322]:
>> ERROR:dialog:dlg_replicated_create: Received malformed UDP binary packet!
>> Dec  1 22:20:58 gcl-ss01a /usr/sbin/opensips[12322]:
>> ERROR:dialog:receive_binary_packet: Failed to process a binary packet!
>> Dec  1 22:20:58 gcl-ss01a kernel: [88582.744765] opensips[12323]:
>> segfault at 10 ip 7fe5dc96d7c7 sp 7fff45469d80 error 4 in
>> dialog.so[7fe5dc949000+49000]
>>
>>
>>  for me this is like if the dialog need to have the same listening
>> information, how should i configure this ?
>>
>>
>> Thanks
>>
>
>
>
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>
>
>
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Re: [OpenSIPS-Users] tests with binary replication

[Mike Tesliuk]

So, nobody have an idea about this ?


2013/12/1 Mike Tesliuk 

> Hello Guys,
>
> Im trying to use the binary interface to test, but when the replication
> happens i get this on log file.
>
> Dec  1 22:20:58 gcl-ss01a /usr/sbin/opensips[12322]:
> ERROR:dialog:dlg_replicated_create: Dialog in DB doesn't match any
> listening sockets
> Dec  1 22:20:58 gcl-ss01a /usr/sbin/opensips[12322]:
> ERROR:dialog:dlg_replicated_create: Received malformed UDP binary packet!
> Dec  1 22:20:58 gcl-ss01a /usr/sbin/opensips[12322]:
> ERROR:dialog:receive_binary_packet: Failed to process a binary packet!
> Dec  1 22:20:58 gcl-ss01a kernel: [88582.744765] opensips[12323]: segfault
> at 10 ip 7fe5dc96d7c7 sp 7fff45469d80 error 4 in
> dialog.so[7fe5dc949000+49000]
>
>
> for me this is like if the dialog need to have the same listening
> information, how should i configure this ?
>
>
> Thanks
>
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[OpenSIPS-Users] tests with binary replication

[Mike Tesliuk]

Hello Guys,

Im trying to use the binary interface to test, but when the replication
happens i get this on log file.

Dec  1 22:20:58 gcl-ss01a /usr/sbin/opensips[12322]:
ERROR:dialog:dlg_replicated_create: Dialog in DB doesn't match any
listening sockets
Dec  1 22:20:58 gcl-ss01a /usr/sbin/opensips[12322]:
ERROR:dialog:dlg_replicated_create: Received malformed UDP binary packet!
Dec  1 22:20:58 gcl-ss01a /usr/sbin/opensips[12322]:
ERROR:dialog:receive_binary_packet: Failed to process a binary packet!
Dec  1 22:20:58 gcl-ss01a kernel: [88582.744765] opensips[12323]: segfault
at 10 ip 7fe5dc96d7c7 sp 7fff45469d80 error 4 in
dialog.so[7fe5dc949000+49000]


for me this is like if the dialog need to have the same listening
information, how should i configure this ?


Thanks
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Re: [OpenSIPS-Users] my opensips can`t registered

[Mike Tesliuk]

According with you maxfwd log you have a loop

Nov 28 15:19:00 MYCALL-500-73
/usr/local/opensips_proxy/sbin/opensips[28994]:
DBG:maxfwd:is_maxfwd_present: value = 70

and if user 342 is not a real user so you have somebody trying to login on
your server


2013/11/28 hualong@busap.com 

>  hello all,
>   my opensips server can`t  registered  and call , I saw this
> information in the log file:
>
>   0-73 /usr/local/opensips_proxy/sbin/opensips[28992]:
> DBG:core:receive_msg: cleaning up
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:parse_msg: SIP
> Request:
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:parse_msg:
> method:  
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:parse_msg:
> uri: 
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:parse_msg:
> version: 
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:parse_headers:
> flags=2
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:parse_via_param:
> found param type 235,  = ; state=6
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:parse_via_param:
> found param type 232,  = ; state=16
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:parse_via: end of
> header reached, state=5
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:parse_headers: via
> found, flags=2
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:parse_headers:
> this is the first via
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:receive_msg: After
> parse_msg...
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:receive_msg:
> preparing to run routing scripts...
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:check_ip_address:
> params 178.162.199.65, 127.0.0.1, 0
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:parse_headers:
> flags=100
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:parse_to: end of
> header reached, state=10
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:parse_to:
> display={"342"}, ruri={sip:342@172.16.100.73}
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:get_hdr_field:
>  [31]; uri=[sip:342@172.16.100.73]
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:get_hdr_field: to
> body ["342" #015#012]
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:get_hdr_field:
> cseq : <1> 
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]:
> DBG:maxfwd:is_maxfwd_present: value = 70
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:uri:has_totag: no totag
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:parse_headers:
> flags=78
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:tm:t_lookup_request:
> start searching: hash=49499, isACK=0
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:tm:matching_3261:
> RFC3261 transaction matching failed
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:tm:t_lookup_request: no
> transaction found
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:parse_headers:
> flags=200
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:get_hdr_field:
> content_length=0
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:get_hdr_field:
> found end of header
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:rr:find_first_route: No
> Route headers found
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:rr:loose_route: There
> is no Route HF
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:grep_sock_info:
> checking if host==us: 13==13 &&  [172.16.100.73] == [172.16.100.73]
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:grep_sock_info:
> checking if port 5060 matches port 5060
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:parse_to: end of
> header reached, state=10
> Nov 28 15:19:00 MYCALL-500-73
> /usr/local/opensips_proxy/sbin/opensips[28994]: DBG:core:parse_to:
> display={"342"}, ruri={sip:342@172.16.100.73}
> Nov 28 15:19:00 MYCA

[OpenSIPS-Users] perl module load

[Mike Tesliuk]

Hello Guys,

I was talking with a opensips users from Brazilian mail list (unofficial)

https://groups.google.com/forum/#!forum/opensipsbrasil


and the guy was trying to use the example from opensips website

http://www.opensips.org/Documentation/Tutorials-Perl-183-to-180

After some problems with dependencies and to load the opensips libs, when
the user try to start opensips the same get a crash, but, when i put the
perl module as the last module to be load the opensips run's ok

So, on the opensips documentation (
http://www.opensips.org/html/docs/modules/1.9.x/perl.html ) on dependencies
section ( http://www.opensips.org/html/docs/modules/1.9.x/perl.html#id248936)
just the sl modules need to be loaded before the perl module.

There is another dependencies that are not on documentation (module to be
loaded ?) , i have the cfg file and the script from the user and i can make
the error, so if you guys want (and send me the instructions (if needed gdb
because i dont know howto use) or another method) i can replicate the error
and send

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Re: [OpenSIPS-Users] Opensips 1.10 NAT

[Mike Tesliuk]

sl_send_reply("403","Forbidden auth ID");
>>> exit;
>>> }
>>>
>>> consume_credentials();
>>> # caller authenticated
>>>
>>> } else {
>>> # if caller is not local, then called number
>>> must be local
>>>
>>> if (!is_uri_host_local()) {
>>> send_reply("403","Rely forbidden");
>>> exit;
>>> }
>>> }
>>>
>>> }
>>>
>>> # preloaded route checking
>>> if (loose_route()) {
>>> xlog("L_ERR",
>>> "Attempt to route with preloaded Route's
>>> [$fu/$tu/$ru/$ci]");
>>> if (!is_method("ACK"))
>>> sl_send_reply("403","Preload Route denied");
>>> exit;
>>> }
>>>
>>> # record routing
>>> if (!is_method("REGISTER|MESSAGE"))
>>> record_route();
>>>
>>> # account only INVITEs
>>> if (is_method("INVITE")) {
>>>
>>> # create dialog with timeout
>>> if ( !create_dialog("B") ) {
>>> send_reply("500","Internal Server Error");
>>> exit;
>>> }
>>>
>>> setflag(ACC_DO); # do accounting
>>> }
>>>
>>>
>>> if (!is_uri_host_local()) {
>>> append_hf("P-hint: outbound\r\n");
>>>
>>> route(relay);
>>> }
>>>
>>> # requests for my domain
>>>
>>> if (is_method("PUBLISH|SUBSCRIBE"))
>>> {
>>> sl_send_reply("503", "Service Unavailable");
>>> exit;
>>> }
>>>
>>> if (is_method("REGISTER"))
>>> {
>>> fix_nated_register();
>>> fix_nated_contact();
>>>     # authenticate the REGISTER requests
>>> if (!www_authorize("", "subscriber"))
>>> {
>>> www_challenge("", "0");
>>> exit;
>>> }
>>>
>>> if (!db_check_to())
>>> {
>>> sl_send_reply("403","Forbidden auth ID");
>>> exit;
>>> }
>>>
>>> if ( 0 ) setflag(TCP_PERSISTENT);
>>>
>>> if (!save("location"))
>>> sl_reply_error();
>>>
>>> exit;
>>> }
>>>
>>> if ($rU==NULL) {
>>> # request with no Username in RURI
>>> sl_send_reply("484","Address Incomplete");
>>> exit;
>>> }
>>>
>>>
>>> # apply DB based aliases
>>> alias_db_lookup("dbaliases");
>>>
>>>
>>> # apply transformations from dialplan table
>>> dp_translate("0","$rU/$rU");
>>>
>>>
>>> if ($rU=~"^\+[1-9][0-9]+$") {
>>>
>>> if (!do_routing("0")) {
>>> send_reply("500","No PSTN Route found");
>>> exit;
>>> }
>>>
>>> route(relay);
>>> exit;
>>> }
>>>
>>>
>>> # do lookup with method filtering
>>> if (!lookup("location","m")) {
>>> if (!db_does_uri_exist()) {
>>> send_reply("420","Bad Extension");
>>> exit;
>>> }
>>>
>>>  t_newtran();
>>> t_reply("404", "Not Found");
>>> exit;
>>> }
>>>
>>> if (isbflagset(NAT)) setflag(NAT);
>>>
>>> # when routing via usrloc, log the missed calls also
>>> setflag(ACC_MISSED);
>>> route(relay);
>>> }
>>>
>>>
>>> route[relay] {
>>> # for INVITEs enable some additional helper routes
>>> if (is_method("INVITE")) {
>>> if(nat_uac_test("127")){
>>> # Usuario identificado como atras de nat
>>> xlog("Usuario atras de nat em handle nat");
>>> fix_nated_contact();
>>> }
>>> if (isflagset(NAT)) {
>>> rtpproxy_offer("ro");
>>> }
>>>
>>> t_on_branch("per_branch_ops");
>>> t_on_reply("handle_nat");
>>> t_on_failure("missed_call");
>>> }
>>>
>>> if (isflagset(NAT)) {
>>> add_rr_param(";nat=yes");
>>> }
>>>
>>> if (!t_relay()) {
>>> send_reply("500","Internal Error");
>>> };
>>> exit;
>>> }
>>>
>>>
>>>
>>>
>>> branch_route[per_branch_ops] {
>>> xlog("new branch at $ru\n");
>>> }
>>>
>>>
>>> onreply_route[handle_nat] {
>>> #   if (nat_uac_test("1"))
>>> #   fix_nated_contact();
>>> #   if ( isflagset(NAT) )
>>> #   rtpproxy_answer("ro");
>>> #   xlog("incoming reply\n");
>>>
>>> # Recebemos resposta do pacote
>>> xlog("incoming reply\n");
>>>
>>> # Verificamos aqui se esta requisicao possui SDP
>>> if(is_method("ACK") && has_body("application/sdp")){
>>> # Atendemos no rtpproxy
>>> rtpproxy_answer();
>>>
>>> }else if(has_body("application/sdp")){
>>> # possuindo sdp vamos re-escrever a informacao
>>> #fix_nated_sdp("2");
>>> rtpproxy_offer();
>>> }
>>>
>>>
>>> # Vamos tentar identificar se o usuario esta atras de nat
>>> # executamos neste nivel pois aqui sera executado
>>> # no momento que recebemos resposta, assim garantimos
>>> # que em todos os casos manipularemos o nat
>>> if(nat_uac_test("127")){
>>> # Usuario identificado como atras de nat
>>> xlog("Usuario atras de nat em handle nat");
>>> fix_nated_contact();
>>> }
>>>
>>> }
>>>
>>>
>>> failure_route[missed_call] {
>>> if (t_was_cancelled()) {
>>> exit;
>>> }
>>>
>>> # uncomment the following lines if you want to block client
>>> # redirect based on 3xx replies.
>>> ##if (t_check_status("3[0-9][0-9]")) {
>>> ##t_reply("404","Not found");
>>> ##  exit;
>>> ##}
>>>
>>>
>>> }
>>>
>>>
>>>
>>> local_route {
>>> if (is_method("BYE") && $DLG_dir=="UPSTREAM") {
>>>
>>> acc_db_request("200 Dialog Timeout", "acc");
>>>
>>> }
>>> }
>>>
>>>
>>>
>>>
>>> Atenciosamente.
>>> Eng.° Rodrigo Ferreira
>>> ITIL v3 Certified
>>>
>>>
>>>
>>>
>>> 2013/10/4 Rodrigo Ferreira 
>>> Yes I did Mike,
>>>
>>> and my SIP messages are ok.
>>>
>>> I will take a look at your tutorial.
>>>
>>> tks
>>>
>>>
>>>
>>> Atenciosamente.
>>> Eng.° Rodrigo Ferreira
>>> ITIL v3 Certified
>>>
>>>
>>>
>>>
>>> 2013/10/3 Mike Tesliuk 
>>> Did you try to made some debug rodrigo ? maybe some rule is missing on
>>> your route script
>>>
>>> i made a tutorial over version 1.9 that you can check
>>>
>>> [portugues]
>>> http://opensips.com.br/wiki/index.php?title=Opensips_NAT_Script_com_RTPproxy
>>> [english]
>>> http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_English
>>>
>>>
>>>
>>>
>>> 2013/10/3 Rodrigo Ferreira 
>>> Hi guys,
>>>
>>> After a long time without using Opensips (almost a year) I tried to
>>> install the opensips 1.10 and everything went well BUT when I make a call,
>>> there's no audio, I know that is something because of NAT, but I have the
>>> nathelper and rtpproxy configuration on my opensips.cfg.
>>>
>>> There's anything else that I could take a look at?
>>>
>>> Thanks
>>>
>>>
>>> Atenciosamente.
>>> Eng.° Rodrigo Ferreira
>>> ITIL v3 Certified
>>>
>>>
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>> ___
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>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> --
> rickygm
>
> http://gnuforever.homelinux.com
>
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Re: [OpenSIPS-Users] Problem with presence_xml module Opensips 1.9

[Mike Tesliuk]

Im Sorry Estefania, when i reply to you i forget about this options




2013/11/25 Estefania Figueroa Buitrago 

> Hi,
>
> I'm very grateful. The server is starting right now without any problems.
> It
> works fine with the following configuration. I hope it could help someone
> else.
>
> -
>
> ### XCAP
> loadmodule "xcap.so"
> modparam("xcap", "db_url",
> "mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
> modparam("xcap", "integrated_xcap_server", 1)
>
>
>  PRESENCE modules
> loadmodule "presence.so"
> loadmodule "presence_xml.so"
> modparam("presence", "db_url",
> "mysql://opensips:opensipsrw@localhost/opensips") # CUSTOMIZE ME
> modparam("presence_xml", "force_active", 1)
> modparam("presence", "server_address", "sip:x.x.x.x:5065") # CUSTOMIZE ME
>
> -
>
> Regards,
> Estefanía
>
>
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/Problem-with-presence-xml-module-Opensips-1-9-tp7588356p7588693.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
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Re: [OpenSIPS-Users] Checking src IP against addresses from DNS?

[Mike Tesliuk]

I think that you should create a DNS cache server in your network, create a
perl script to check everything and use this on opensips, i think that
should work well


2013/11/22 Daniel Nihlén 

> Seems it does not resolve srv, but its works for a-names.
>
> Thanks.
> Daniel
>
>
>
> 2013/11/21 Mike Tesliuk 
>
>> http://www.opensips.org/Documentation/Script-Tran#toc63
>>
>>
>> 2013/11/21 Daniel Nihlén 
>>
>>>  Hi,
>>>
>>> Any way to have openSIPS check source IP according to dns.
>>>
>>> Now for example:
>>> if(src_ip==10.10.10.10 || src_ip==11.11.11.11)
>>>
>>> I would like:
>>> if(src_ip==parterfqdn.com)
>>>
>>> and source ip to be compared with any ip available from dns for
>>> partnerfqdn.com, a-record, srv for different transport etc.
>>>
>>> Does any one know if there is any built in function to achieve this? Or
>>> a recommended solution.
>>>
>>> Thanks
>>> Daniel Nihlén
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
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>
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Re: [OpenSIPS-Users] Problem with presence_xml module Opensips 1.9

[Mike Tesliuk]

I made this configuration/howto and work fine to me without xcap, using the
version 1.9 i think that was 2 months ago


2013/11/21 Estefania Figueroa Buitrago 

> Hi Mike. Basically I have the same configuration, the problem is, like I
> said
> before, that it doesn't matter if I have set the force_active param to one
> because I get an error, in which the system asks for a xcap server.
>
> This error comes from older versions but is still there :(. Nevertheless
> I'm
> gonna check the tutorial (and use for sure Google Translate :P). They could
> have a different approach and make it work.
>
> So thanks, I will let you know if it works.
>
> Regards,
> Estefanía
>
>
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/Problem-with-presence-xml-module-Opensips-1-9-tp7588356p7588614.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
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Re: [OpenSIPS-Users] Problem with presence_xml module Opensips 1.9

[Mike Tesliuk]

Hello Estefania,

You can check this tutorial (you have the complete cfg file on the end)

http://opensips.com.br/wiki/index.php?title=Opensips_1.9

There you have a working presence without xcap as you want

but basicaly

 PRESENCE modules
loadmodule "presence.so"
loadmodule "presence_xml.so"
modparam("presence", "db_url",
"mysql://opensips:_SUA_SENHA_AQUI_@localhost/opensips") # CUSTOMIZE ME
modparam("presence_xml", "force_active", 1)
modparam("presence", "server_address", "sip:127.0.0.1:5060") # CUSTOMIZE ME



After this you just need handle the request
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Re: [OpenSIPS-Users] Checking src IP against addresses from DNS?

[Mike Tesliuk]

http://www.opensips.org/Documentation/Script-Tran#toc63


2013/11/21 Daniel Nihlén 

>  Hi,
>
> Any way to have openSIPS check source IP according to dns.
>
> Now for example:
> if(src_ip==10.10.10.10 || src_ip==11.11.11.11)
>
> I would like:
> if(src_ip==parterfqdn.com)
>
> and source ip to be compared with any ip available from dns for
> partnerfqdn.com, a-record, srv for different transport etc.
>
> Does any one know if there is any built in function to achieve this? Or a
> recommended solution.
>
> Thanks
> Daniel Nihlén
>
> ___
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> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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Re: [OpenSIPS-Users] Problem with Opensips-cp CDR And Users registration

[Mike Tesliuk]

you should load this modules on opensips.cfg and configure to work


2013/11/15 Vishnu Vardhan 

> Hi,
>
> I installed opensips 1.8 with opensips 5.0.And i integrate with it
> asterisk 11.0.3.When i tried to register the users it is not updating
> in cdr-viewer and in softphone also users are not registering after
> registering in users pannel.Can any one help he to get out of this
> problem.And pls see the below log message of opensips which i got.
>
> Nov 15 05:00:27 developer-asterisk-vm24
> /usr/local/sbin/opensips[3373]: ERROR:mi_fifo:mi_fifo_server: command
> sip_trace is not available
>
> Nov 15 05:01:03 developer-asterisk-vm24
> /usr/local/sbin/opensips[3373]: ERROR:mi_fifo:mi_fifo_server: command
> dr_gw_status is not available
>
> Nov 15 05:01:30 developer-asterisk-vm24
> /usr/local/sbin/opensips[3373]: ERROR:mi_fifo:mi_fifo_server: command
> dlg_list is not available
>
> Nov 15 05:09:37 developer-asterisk-vm24
> /usr/local/sbin/opensips[3373]: ERROR:mi_fifo:mi_fifo_server: command
> sip_trace is not available
>
> Nov 15 05:09:58 developer-asterisk-vm24
> /usr/local/sbin/opensips[3373]: ERROR:mi_fifo:mi_fifo_server: command
> dr_gw_status is not available
>
> Nov 15 05:10:02 developer-asterisk-vm24
> /usr/local/sbin/opensips[3373]: ERROR:mi_fifo:mi_fifo_server: command
> dlg_list is not available
>
> Nov 15 06:08:45 developer-asterisk-vm24
> /usr/local/sbin/opensips[3373]: ERROR:mi_fifo:mi_fifo_server: command
> domain_reload is not available
>
> Nov 15 06:12:23 developer-asterisk-vm24
> /usr/local/sbin/opensips[3373]: ERROR:mi_fifo:mi_fifo_server: command
> domain_reload is not available
>
> Regards,
> Vishnu
>
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Re: [OpenSIPS-Users] DR and failover

[Mike Tesliuk]

on failure route you should use the function use_next_gw

 if (use_next_gw()) {
route(4);
}


you have a tutorial on opensips website

http://www.opensips.org/Documentation/Tutorials#toc2




2013/11/15 Nick Cameo 

> Hello Everyone,
>
> We use the DR module and LOVE IT! Quick question regarding failing
> over to the next gateway. Do we need to do anything in the failure
> route of our script? Or is it just as simple as entering the gateway
> in dr_rules.gwlist (ie, 1,2,3.), and having the entry in
> dr_gatways?
>
> Kind Regards,
>
> Nick.
>
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Re: [OpenSIPS-Users] Opensips 1.10 NAT

[Mike Tesliuk]

You can use the avpops module

avp_db_load("$fu","$avp(678)");

This will load the preferences from the user on avp(678) , so you
check the value and force the use of rtpproxy/mediaproxy as you do
when the user is behind proxy.

you can use the avp_db_query also

avp_db_query("select value from usr_preferences where username='$fu'
and atrribute = 'USE_NAT'",
"$avp(678)");

in this case you should have an information for this username ($fu)
using an attribute USE_NAT , you can set the
value to 1 or 0 , and you do the rest on you dialplan





2013/11/5 troxlinux 

> Thnk mike,  an example where I can watch this?
>
>
> 2013/11/5 Mike Tesliuk 
>
>> You can set a flag on usr_preferences to force the nat to that customer
>> so you can manage this on your dialplan
>>
>> if the user cannot be recognized over nat help you can force.
>>
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>>
>>
>
>
> --
> rickygm
>
> http://gnuforever.homelinux.com
>
> ___
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>
>
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Re: [OpenSIPS-Users] Opensips 1.10 NAT

[Mike Tesliuk]

You can set a flag on usr_preferences to force the nat to that customer so
you can manage this on your dialplan

if the user cannot be recognized over nat help you can force.
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Re: [OpenSIPS-Users] Siremis with Opensips - possible ?

[Mike Tesliuk]

Yep,

I will create a test enviroment to make this easy, as to inform about
problem and also new ideas .


2013/10/30 Bogdan-Andrei Iancu 

> **
> Reporting bugs or lack of functionality may be a good start in doing the
> CP better - we are open and willing to do the coding.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
>
> On 10/30/2013 03:18 PM, Mike Tesliuk wrote:
>
>  Yep,
>
> This is a very good sugestion :)
>
>  I like OpenSIPS-CP , on my instalation i see some errors that are not a
> problem for me but can be for other users, but we realy can make some
> contribuition to this project.
>
>  Im not a development dedicated guy, but i know a a litle bit of a lot of
> things , this could be very nice
>
>
> 2013/10/30 Bogdan-Andrei Iancu 
>
>> Hello Ewgeny,
>>
>> Siremis is not designed for OpenSIPS and you may have huge compatibility
>> issues - of course, if you want to dive into this, feel free :).
>>
>> Maybe a better approach (for you and for the rest of the community) is
>> to identify things that should be improved in OpenSIPS Control Panel and
>> let's make it better.
>>
>> Best regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developer
>> http://www.opensips-solutions.com
>>
>>
>> On 10/28/2013 04:11 PM, Ewgeny wrote:
>> > Hello!
>> >
>> > Is that possible to use Siremis WEB GUI
>> > http://siremis.asipto.com/2013/05/08/siremis-v4-0-0-released/ with
>> > Opensips ?
>> > Siremis has better features for SIP Database management then Opensips
>> > Control Panel, and lots of other feature that Opensip CP does'nt have.
>> > I tried to install version 3.2 in conjunction with Opensips, but it
>> > does not work well, a lot of errors.
>> > Maybe someone has experience with Siremis and Opensips adaptation?
>> >
>> >
>> > Best Regards,
>> > Ewgeny
>> >
>> >
>> >
>> >
>> > ___
>> > Users mailing list
>> > Users@lists.opensips.org
>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> >
>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
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>
>
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Re: [OpenSIPS-Users] Siremis with Opensips - possible ?

[Mike Tesliuk]

Yep,

This is a very good sugestion :)

I like OpenSIPS-CP , on my instalation i see some errors that are not a
problem for me but can be for other users, but we realy can make some
contribuition to this project.

Im not a development dedicated guy, but i know a a litle bit of a lot of
things , this could be very nice


2013/10/30 Bogdan-Andrei Iancu 

> Hello Ewgeny,
>
> Siremis is not designed for OpenSIPS and you may have huge compatibility
> issues - of course, if you want to dive into this, feel free :).
>
> Maybe a better approach (for you and for the rest of the community) is
> to identify things that should be improved in OpenSIPS Control Panel and
> let's make it better.
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
>
> On 10/28/2013 04:11 PM, Ewgeny wrote:
> > Hello!
> >
> > Is that possible to use Siremis WEB GUI
> > http://siremis.asipto.com/2013/05/08/siremis-v4-0-0-released/ with
> > Opensips ?
> > Siremis has better features for SIP Database management then Opensips
> > Control Panel, and lots of other feature that Opensip CP does'nt have.
> > I tried to install version 3.2 in conjunction with Opensips, but it
> > does not work well, a lot of errors.
> > Maybe someone has experience with Siremis and Opensips adaptation?
> >
> >
> > Best Regards,
> > Ewgeny
> >
> >
> >
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
>
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Re: [OpenSIPS-Users] Trying to change the $ru to an address in location

[Mike Tesliuk]

Ok, the problem is caused by an engage_rtp_proxy() , i remove this options
and everything seems to work now.


2013/9/25 Mike Tesliuk 

> I dont know exactly why ( im trying to find this why ), if i after check
> te $ru immediatly call the lookup , and send to the relay the package reach
> the destination , i will now investigate which rule cause the problem.
>
> if you guys find something on the log that i send please tell me
>
>
> 2013/9/25 Mike Tesliuk 
>
>> Hello Vlad, first of all thanks for your reply .
>>
>>
>> Below some information that i hope you can check , in advice i say that
>> the package dont go out from opensips, i see on the debug, but i dont see
>> the package on my router (near opensips)
>>
>> i change the ip address information below to this
>>
>>
>>
>> ___ASTERISK___DESTINATION___ = the final destination of the call
>>
>> OPENSIPS___IP___ = the opensips ip address
>>
>> ___ASTERISK_ORIGINATE = the asterisk that originate the call to
>> opensips
>>
>>
>>
>> Just to remember the case, i receive on opensips a call  did@opensips_ip, i 
>> check everything , and change the $ru to account@opensips_ip
>> and send to lookup(location), the user was find and the send to t_relay
>>
>> I will apreciate a lot if you or someone else can check the logs below
>> because i can't figure out why this is not working, i dont see any relevant
>> message on log about the packege, everything seems to be executed, the only
>> strange thing that i check is this information (
>>
>> Sep 25 21:22:19 [25420] DBG:tm:matching_3261: RFC3261 transaction
>> matching failed
>> Sep 25 21:22:19 [25420] DBG:tm:t_lookup_request: no transaction found
>> Sep 25 21:22:19 [25420] DBG:tm:run_reqin_callbacks: trans=0x7fd613156710,
>> callback type 1, id 2 entered
>> Sep 25 21:22:19 [25420] DBG:core:parse_headers: flags=78
>> Sep 25 21:22:19 [25420] DBG:dialog:new_dlg_val: inserting
>> =
>> Sep 25 21:22:19 [25420] DBG:rr:is_direction: param ftag not found
>>
>>
>> )
>>
>>
>> im using for this test an opensips 1.9.1
>>
>>
>> thanks in advice for any help that you guys can provide.
>>
>>
>>
>> opensipsctl ul show
>>
>> Domain:: location table=512 records=1
>> AOR:: 05501139501010
>> Contact:: sip:s@___ASTERISK___DESTINATION___:5060 Q=
>> Expires:: 94
>> Callid:: 2dd8bd5d6570d1a47caa8ec67ae4197b@
>> ___ASTERISK___DESTINATION___
>> Cseq:: 103
>> User-agent:: PBXUltranet
>>
>> State:: CS_SYNC
>> Flags:: 0
>> Cflag:: 0
>> Socket:: udp:OPENSIPS___IP___:5060
>> Methods:: 4294967295
>>
>>
>>
>> iptables -L -n
>>
>> Chain INPUT (policy ACCEPT)
>> target prot opt source   destination
>>
>> Chain FORWARD (policy ACCEPT)
>> target prot opt source   destination
>>
>> Chain OUTPUT (policy ACCEPT)
>> target prot opt source   destination
>>
>>
>>
>>
>> - tcpdump on opensips --
>>
>>
>> 21:17:51.194560 16:44:c7:4e:35:2d > 00:30:48:94:6d:b6, ethertype IPv4
>> (0x0800), length 1261: (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto
>> UDP (17), length 1247)
>> OPENSIPS___IP___.5060 > ___ASTERISK___DESTINATION___.5060: [bad
>> udp cksum fe56!] SIP, length: 1219
>> INVITE sip:s@___ASTERISK___DESTINATION___:5060 SIP/2.0
>>
>>
>>
>>
>> on my router i don't see any package, on the debug i have the result below
>>
>>
>>
>> Sep 25 21:22:19 [25420] DBG:core:parse_msg: SIP Request:
>> Sep 25 21:22:19 [25420] DBG:core:parse_msg:  method:  
>> Sep 25 21:22:19 [25420] DBG:core:parse_msg:  uri: > OPENSIPS___IP___>
>> Sep 25 21:22:19 [25420] DBG:core:parse_msg:  version: 
>> Sep 25 21:22:19 [25420] DBG:core:parse_headers: flags=2
>> Sep 25 21:22:19 [25420] DBG:core:parse_via_param: found param type 232,
>>  = ; state=6
>> Sep 25 21:22:19 [25420] DBG:core:parse_via_param: found param type 235,
>>  = ; state=17
>> Sep 25 21:22:19 [25420] DBG:core:parse_via: end of header reached, state=5
>> Sep 25 21:22:19 [25420] DBG:core:parse_headers: via found, flags=2
>> Sep 25 21:22:19 [25420] DBG:core:parse_headers: this is the first via
>> Sep 25 21:22:19 [25420] DBG:core:receive_msg: After parse_msg...
>> Sep 25 21:22:19 [2

Re: [OpenSIPS-Users] FW: Audio and Video not working for different otuside netrwork

[Mike Tesliuk]

 if i say something wrong please somebody correct my 


Hello Rajesh,

You are using the nat_uac_test with parameter 23, this means parameters 16,
4, 2, 1 , what means


   -

   *1* - Contact header field is searched for occurrence of RFC1918
   addresses.
   -

   *2* - the "received" test is used: address in Via is compared against
   source IP address of signaling
   -

   *4* - Top Most VIA is searched for occurrence of RFC1918 addresses
   -

   *16* - test if the source port is different from the port in Via



i dont know if you understand but this is a binary count, you can check in
this way

0010111 -> this is what you turn on

in this case, if your package does not contains an Private ip address on
contact header, or does not contains a received on VIA different from the
ip address of the signalling, does not contais on VIA an private ip address
and the source port is not different from port on VIA , so your rule will
not match (just on match is enought)

Look at this invite below (sended from a zoiper)

 204.16.0.26:60340 -> 204.16.1.50:5060
INVITE sip:101@204.16.1.50;transport=UDP SIP/2.0.
Via: SIP/2.0/UDP 75.74.203.73:60340
;branch=z9hG4bK-d8754z-f6a3eadc786e7359-1---d8754z-;rport.
Max-Forwards: 70.
Contact: .
To: .
From: "102";tag=489f8f45.
Call-ID: ZGNhYTQzNjIyOGFkYWNhOWQ3ZmQ2ZDVkYjhiNGI4MGE..
CSeq: 1 INVITE.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
SUBSCRIBE.
Content-Type: application/sdp.
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri.
User-Agent: Zoiper Communicator 2.04.10164 rev.10204.
Allow-Events: presence, kpml.
Content-Length: 352.


you can see the ip on signalling coming from 204.16.0.26 port 60340
on via you have 75.74.203.73:60340, so  you have a different ip address
from signalling or via , in this case you will set the NAT variable, but
check the invite below.

#
U 204.16.0.26:5062 -> 204.16.1.50:5060
INVITE sip:102@204.16.1.50 SIP/2.0.
Via: SIP/2.0/UDP 204.16.0.26:5062;branch=z9hG4bK1527256431.
From: "Mike" ;tag=1050377705.
To: .
Call-ID: 83821284@10.254.254.6.
CSeq: 1 INVITE.
Contact: .
Content-Type: application/sdp.
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER,
SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE.
Max-Forwards: 70.
User-Agent: Yealink SIP-T20P 9.70.0.121.
Supported: replaces.
Allow-Events: talk,hold,conference,refer,check-sync.
Content-Length: 304

You have the same port on signalling and on VIA, in this case the rule will
no match and variable will not be set and this is a phone behind a nat


so, you should try to remove the if where you call the rtpproxy offer and
answer (just for test purpose)

you should increment you debug info too

/* uncomment the following lines to enable debugging */
#debug=6
#fork=no
#log_stderror=yes







2013/9/28 Rajesh Babu 

> Hi,
>
> ** **
>
>I have attached the logs and my routing file @
> http://pastebin.com/hu0bQGVw
>
> ** **
>
> Please help me out in nailing this.
>
> ** **
>
> Thanks 
>
> Rajesh
>
> ** **
>
> *From:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *On Behalf Of *Mike Tesliuk
> *Sent:* Friday, 27 September, 2013 11:25 PM
>
> *To:* OpenSIPS users mailling list
> *Subject:* Re: [OpenSIPS-Users] FW: Audio and Video not working for
> different otuside netrwork
>
> ** **
>
> If possible, paste your route file too
>
> ** **
>
> 2013/9/27 Mike Tesliuk 
>
> start your opensips in debug mode, try to make the call, get all the
> message and paste in some pastebin website and show us the link
>
> ** **
>
> 2013/9/27 Rajesh Babu 
>
> I am getting Error 483, too many Hops, There is no other error messages i
> am getting. Please some one help me out in this
>
>  
>
> *From:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *On Behalf Of *Rajesh Babu
> *Sent:* Friday, 27 September, 2013 6:08 PM
>
>
> *To:* 'OpenSIPS users mailling list'
> *Subject:* Re: [OpenSIPS-Users] FW: Audio and Video not working for
> different otuside netrwork
>
>  
>
> HI Mike,
>
>  
>
>Now the RTP is up and i am getting this message on my logs
>
> [root@centos64 rtpproxy-1.2.0]# tailf /var/log/messages
>
> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17800]:
> INFO:core:probe_max_sock_buff: using rcv buffer of 448 kb
>
> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17810]:
> INFO:rtpproxy:rtpp_test: rtp proxy  found, support
> for it enabled
>
> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17812]:
> INFO:rtpproxy:rtpp_test: rtp proxy  found, support
> for it enabled
>
> Sep 28 01:59:4

Re: [OpenSIPS-Users] FW: Audio and Video not working for different otuside netrwork

[Mike Tesliuk]

Take a look over this howto

http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_English


2013/9/28 Mike Tesliuk 

>
>  if i say something wrong please somebody correct my 
>
>
> Hello Rajesh,
>
> You are using the nat_uac_test with parameter 23, this means parameters
> 16, 4, 2, 1 , what means
>
>
>-
>
>*1* - Contact header field is searched for occurrence of RFC1918
>addresses.
>-
>
>*2* - the "received" test is used: address in Via is compared against
>source IP address of signaling
>-
>
>*4* - Top Most VIA is searched for occurrence of RFC1918 addresses
>-
>
>*16* - test if the source port is different from the port in Via
>
>
>
> i dont know if you understand but this is a binary count, you can check in
> this way
>
> 0010111 -> this is what you turn on
>
> in this case, if your package does not contains an Private ip address on
> contact header, or does not contains a received on VIA different from the
> ip address of the signalling, does not contais on VIA an private ip address
> and the source port is not different from port on VIA , so your rule will
> not match (just on match is enought)
>
> Look at this invite below (sended from a zoiper)
>
>  204.16.0.26:60340 -> 204.16.1.50:5060
> INVITE sip:101@204.16.1.50;transport=UDP SIP/2.0.
> Via: SIP/2.0/UDP 75.74.203.73:60340
> ;branch=z9hG4bK-d8754z-f6a3eadc786e7359-1---d8754z-;rport.
> Max-Forwards: 70.
> Contact: .
> To: .
> From: "102";tag=489f8f45.
> Call-ID: ZGNhYTQzNjIyOGFkYWNhOWQ3ZmQ2ZDVkYjhiNGI4MGE..
> CSeq: 1 INVITE.
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
> SUBSCRIBE.
> Content-Type: application/sdp.
> Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri.
> User-Agent: Zoiper Communicator 2.04.10164 rev.10204.
> Allow-Events: presence, kpml.
> Content-Length: 352.
>
>
> you can see the ip on signalling coming from 204.16.0.26 port 60340
> on via you have 75.74.203.73:60340, so  you have a different ip address
> from signalling or via , in this case you will set the NAT variable, but
> check the invite below.
>
> #
> U 204.16.0.26:5062 -> 204.16.1.50:5060
> INVITE sip:102@204.16.1.50 SIP/2.0.
> Via: SIP/2.0/UDP 204.16.0.26:5062;branch=z9hG4bK1527256431.
> From: "Mike" ;tag=1050377705.
> To: .
> Call-ID: 83821284@10.254.254.6.
> CSeq: 1 INVITE.
> Contact: .
> Content-Type: application/sdp.
> Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER,
> SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE.
> Max-Forwards: 70.
> User-Agent: Yealink SIP-T20P 9.70.0.121.
> Supported: replaces.
> Allow-Events: talk,hold,conference,refer,check-sync.
> Content-Length: 304
>
> You have the same port on signalling and on VIA, in this case the rule
> will no match and variable will not be set and this is a phone behind a nat
>
>
> so, you should try to remove the if where you call the rtpproxy offer and
> answer (just for test purpose)
>
> you should increment you debug info too
>
> /* uncomment the following lines to enable debugging */
> #debug=6
> #fork=no
> #log_stderror=yes
>
>
>
>
>
>
>
> 2013/9/28 Rajesh Babu 
>
>> Hi,
>>
>> ** **
>>
>>I have attached the logs and my routing file @
>> http://pastebin.com/hu0bQGVw
>>
>> ** **
>>
>> Please help me out in nailing this.
>>
>> ** **
>>
>> Thanks ****
>>
>> Rajesh
>>
>> ** **
>>
>> *From:* users-boun...@lists.opensips.org [mailto:
>> users-boun...@lists.opensips.org] *On Behalf Of *Mike Tesliuk
>> *Sent:* Friday, 27 September, 2013 11:25 PM
>>
>> *To:* OpenSIPS users mailling list
>> *Subject:* Re: [OpenSIPS-Users] FW: Audio and Video not working for
>> different otuside netrwork
>>
>> ** **
>>
>> If possible, paste your route file too
>>
>> ** **
>>
>> 2013/9/27 Mike Tesliuk 
>>
>> start your opensips in debug mode, try to make the call, get all the
>> message and paste in some pastebin website and show us the link
>>
>> ** **
>>
>> 2013/9/27 Rajesh Babu 
>>
>> I am getting Error 483, too many Hops, There is no other error messages i
>> am getting. Please some one help me out in this
>>
>>  
>>
>> *From:* users-boun...@lists.opensips.org [mailto:
>> users-boun...@lists.opensips.org] *On Behalf Of *Rajesh Babu
>> *Sent:* Friday, 27 September, 2013 6:08 PM
>>
>>
>> *T

Re: [OpenSIPS-Users] Trying to change the $ru to an address in location

[Mike Tesliuk]

I dont know exactly why ( im trying to find this why ), if i after check te
$ru immediatly call the lookup , and send to the relay the package reach
the destination , i will now investigate which rule cause the problem.

if you guys find something on the log that i send please tell me


2013/9/25 Mike Tesliuk 

> Hello Vlad, first of all thanks for your reply .
>
>
> Below some information that i hope you can check , in advice i say that
> the package dont go out from opensips, i see on the debug, but i dont see
> the package on my router (near opensips)
>
> i change the ip address information below to this
>
>
>
> ___ASTERISK___DESTINATION___ = the final destination of the call
>
> OPENSIPS___IP___ = the opensips ip address
>
> ___ASTERISK_ORIGINATE = the asterisk that originate the call to
> opensips
>
>
>
> Just to remember the case, i receive on opensips a call  did@opensips_ip, i 
> check everything , and change the $ru to account@opensips_ip
> and send to lookup(location), the user was find and the send to t_relay
>
> I will apreciate a lot if you or someone else can check the logs below
> because i can't figure out why this is not working, i dont see any relevant
> message on log about the packege, everything seems to be executed, the only
> strange thing that i check is this information (
>
> Sep 25 21:22:19 [25420] DBG:tm:matching_3261: RFC3261 transaction matching
> failed
> Sep 25 21:22:19 [25420] DBG:tm:t_lookup_request: no transaction found
> Sep 25 21:22:19 [25420] DBG:tm:run_reqin_callbacks: trans=0x7fd613156710,
> callback type 1, id 2 entered
> Sep 25 21:22:19 [25420] DBG:core:parse_headers: flags=78
> Sep 25 21:22:19 [25420] DBG:dialog:new_dlg_val: inserting
> =
> Sep 25 21:22:19 [25420] DBG:rr:is_direction: param ftag not found
>
>
> )
>
>
> im using for this test an opensips 1.9.1
>
>
> thanks in advice for any help that you guys can provide.
>
>
>
> opensipsctl ul show
>
> Domain:: location table=512 records=1
> AOR:: 05501139501010
> Contact:: sip:s@___ASTERISK___DESTINATION___:5060 Q=
> Expires:: 94
> Callid:: 2dd8bd5d6570d1a47caa8ec67ae4197b@
> ___ASTERISK___DESTINATION___
> Cseq:: 103
> User-agent:: PBXUltranet
>
> State:: CS_SYNC
> Flags:: 0
> Cflag:: 0
> Socket:: udp:OPENSIPS___IP___:5060
> Methods:: 4294967295
>
>
>
> iptables -L -n
>
> Chain INPUT (policy ACCEPT)
> target prot opt source   destination
>
> Chain FORWARD (policy ACCEPT)
> target prot opt source   destination
>
> Chain OUTPUT (policy ACCEPT)
> target prot opt source   destination
>
>
>
>
> - tcpdump on opensips --
>
>
> 21:17:51.194560 16:44:c7:4e:35:2d > 00:30:48:94:6d:b6, ethertype IPv4
> (0x0800), length 1261: (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto
> UDP (17), length 1247)
> OPENSIPS___IP___.5060 > ___ASTERISK___DESTINATION___.5060: [bad
> udp cksum fe56!] SIP, length: 1219
> INVITE sip:s@___ASTERISK___DESTINATION___:5060 SIP/2.0
>
>
>
>
> on my router i don't see any package, on the debug i have the result below
>
>
>
> Sep 25 21:22:19 [25420] DBG:core:parse_msg: SIP Request:
> Sep 25 21:22:19 [25420] DBG:core:parse_msg:  method:  
> Sep 25 21:22:19 [25420] DBG:core:parse_msg:  uri:  OPENSIPS___IP___>
> Sep 25 21:22:19 [25420] DBG:core:parse_msg:  version: 
> Sep 25 21:22:19 [25420] DBG:core:parse_headers: flags=2
> Sep 25 21:22:19 [25420] DBG:core:parse_via_param: found param type 232,
>  = ; state=6
> Sep 25 21:22:19 [25420] DBG:core:parse_via_param: found param type 235,
>  = ; state=17
> Sep 25 21:22:19 [25420] DBG:core:parse_via: end of header reached, state=5
> Sep 25 21:22:19 [25420] DBG:core:parse_headers: via found, flags=2
> Sep 25 21:22:19 [25420] DBG:core:parse_headers: this is the first via
> Sep 25 21:22:19 [25420] DBG:core:receive_msg: After parse_msg...
> Sep 25 21:22:19 [25420] DBG:core:receive_msg: preparing to run routing
> scripts...
> Sep 25 21:22:19 [25420] DBG:core:parse_headers: flags=
> Sep 25 21:22:19 [25420] DBG:core:parse_to: end of header reached, state=10
> Sep 25 21:22:19 [25420] DBG:core:parse_to: display={},
> ruri={sip:551133992377@OPENSIPS___IP___}
> Sep 25 21:22:19 [25420] DBG:core:get_hdr_field:  [32];
> uri=[sip:551133992377@OPENSIPS___IP___]
> Sep 25 21:22:19 [25420] DBG:core:get_hdr_field: to body [ OPENSIPS___IP___>
> ]
> Sep 25 21:22:19 [25420] DBG:core:get_hdr_field: cseq : <102> 
> Sep 25 21:22:19 [25420] DBG:core:g

Re: [OpenSIPS-Users] Trying to change the $ru to an address in location

[Mike Tesliuk]

0)
Sep 25 21:22:23 [25423] DBG:tm:retransmission_handler:
retransmission_handler : done
Sep 25 21:22:25 [25423] DBG:tm:timer_routine: timer
routine:0,tl=0x7fd613156960 next=(nil), timeout=15
Sep 25 21:22:25 [25423] DBG:tm:final_response_handler: Cancel sent out,
sending 408 (0x7fd613156710)
Sep 25 21:22:25 [25423] DBG:tm:t_should_relay_response: T_code=100,
new_code=408
Sep 25 21:22:25 [25423] DBG:tm:t_pick_branch: picked branch 0, code 408
(prio=800)
Sep 25 21:22:25 [25423] DBG:tm:is_3263_failure: dns-failover test:
branch=0, last_recv=408, flags=1
Sep 25 21:22:25 [25423] DBG:tm:t_should_relay_response: trying DNS-based
failover
Sep 25 21:22:25 [25423] DBG:tm:run_trans_callbacks: trans=0x7fd613156710,
callback type 32, id 2 entered




2013/9/25 Vlad Paiu 

>  Hello,
>
> The approach you're taking seems good, and it should definitely work.
>
> Do you receive any errors in the OpenSIPS logs ? If you make a SIP trace (
> ngrep / tcpdump ) on the OpenSIPS machine, do you see the INVITE message
> getting out of OpenSIPS ?
>
> If you don't see the package being sent out even on the OpenSIPS machine,
> then it means that somehow your script doesn't forward that request, but
> drops it ( because ngrep on the OpenSIPS machine excludes any firewall
> issues - ngrep gets in even before the local iptables output chain ).
>
> Best Regards,
>
> Vlad Paiu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 25.09.2013 15:36, Mike Tesliuk wrote:
>
>  Somebody have at least an idea if this is suposed to work ?
>
>  the package come to me, i check the did, change the $ru, and send to the
> location, the location find the user and the package dont reach the other
> side.
>
>  Thanks
>
>
> 2013/9/22 Mike Tesliuk 
>
>>Hello Guys,
>>
>>
>> Im trying to implement a system to manipulate DID's, the forward for
>> external address is ok, but in trying now to do the same with  a user that
>> is registered
>>
>>
>>  i create a new table where i have the did, the account and the
>> destination, if the destination is null so the opensips will check the
>> account on the location table.
>>
>>  basically i have this
>>
>> if(!$avp(91)){
>> xlog("Did nao encontrado");
>> sl_send_reply("404", "Not Found");
>> exit;
>> }else{
>> xlog("Did encontrado, seguindo
>> regras para utilizacao em location");
>> $ru = "sip:" + $avp(91) +
>> "@IP_ADDRESS:5060";
>> xlog("Novo destino $ru");
>> }
>>
>>
>>  the avp(91) is the user account, the same that the user use to register
>>
>>
>>  when in this situation, the call go trouhg the location module, and the
>> system find the correct address.
>>
>>  i have on the log the same information that i have on opensipsctl ul show
>>
>> Fazendo relay  - sip:05501139501010@IP_ADDR
>> :5081;rinstance=0f9054bc313f0cf1;transport=UDP
>>
>>
>>  below the output from ul show
>>
>> AOR:: 05501139501010
>> Contact:: 
>> sip:05501139501010@IP_ADDR:5081;rinstance=0f9054bc313f0cf1;transport=UDP
>> Q=
>> Expires:: 525
>> Callid:: MTZhNzE1ZDYzYWU4Y2ViZDMzZTQzZWU1N2M0ZGFiZjQ.
>> Cseq:: 2
>> User-agent:: Zoiper Communicator 2.04.10164 rev.10204
>> State:: CS_SYNC
>> Flags:: 0
>> Cflag:: 0
>> Socket:: udp:GW_IP_ADDR:5060
>> Methods:: 5951
>>
>>
>>
>>  But when i make the call, the ngrep show me the send of the invite, but
>> i dont see nothing on the other side.
>>
>>
>>  Below you have the invite
>>
>> U GW_IP_ADDR:5060 -> CUSTOMER_IP_ADDR:5081
>> INVITE 
>> sip:05501139501010@CUSTOMER_IP_ADDR:5081;rinstance=0f9054bc313f0cf1;transport=UDP
>> SIP/2.0.
>> Record-Route: .
>> Via: SIP/2.0/UDP GW_IP_ADDR:5060;branch=z9hG4bK485d.8be72863.0.
>> Via: SIP/2.0/UDP
>> CALLER_IP_ADDR:5060;received=CALLER_IP_ADDR;branch=z9hG4bK1f912c35;rport=5060.
>> Max-Forwards: 69.
>> From: "testemike" ;tag=as657116d5.
>> To: .
>> Contact: .
>> Call-ID: 40d32e5b3c52c58646d996d871ad8471@CALLER_IP_ADDR:5060.
>> CSeq: 102 INVITE.
>> User-Agent: SIP.Ultra

Re: [OpenSIPS-Users] Opensip as transaprent inbound proxy for asterisk

[Mike Tesliuk]

check this resource

http://www.opensips.org/Documentation/Script-CoreFunctions#toc43




2013/10/14 Brett Nemeroff 

> I believe the suggestion is to:
> 1. Allow all calls from OpenSIPs to hit your dial plan (insecure=invite)
> 2. In the dial plan, check for the existence of a custom header, this
> customer header should be inserted by opensips to indicate the original IP
> (append_hf("X-Original-IP: $si");)
> 3. "do something" based on the original IP received.
>
> -Brett
>
>
> --
> Brett Nemeroff
> Sent with Airmail 
>
> On October 14, 2013 at 1:02:09 PM, bluerain 
> (frank21...@yahoo.com)
> wrote:
>
> thx for the suggestion, I don't think asterisk reads the IP from any of
> the
> header or in any part of SIP message. I think asterisk read the IP from
> the
> IP at the network layer. anyhow, if you like you can read my reply to
> Mike.
>
> Thx again. Have you ever encounter the usr_loc module that stop updating
> the DB location table after few hours from initial boot?
>
>
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/Opensip-as-transaprent-inbound-proxy-for-asterisk-tp7588047p7588083.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
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

Re: [OpenSIPS-Users] Write CDR to SYSLOG when mysql statement fail

[Mike Tesliuk]

you should write on both, you decide when generate the acc information
setting the flag, but i dont know if you can receive some variable with the
return of the query over the config file.

you can make your mysql better using failover , replication and everything
, you should fix your problem with your database not  on opensips


2013/10/11 Wilmar Campos 

> Thank you Mike, but how do I control to write only when Mysql insert fails?
> I don't want to end up with every record written to the syslog.
>
> Thanks,
>
> Wilmar
>
>
> On Fri, Oct 11, 2013 at 8:12 AM, Mike Tesliuk  wrote:
>
>> you can use both , just se the log flag (
>> http://www.opensips.org/html/docs/modules/1.9.x/acc#id293023 ) and
>> log_db_flag together (the same for log_missed_flag )
>>
>>
>> 2013/10/11 Wilmar Campos 
>>
>>>  Hi All,
>>>
>>> I hope everybody is good!!
>>> I am writting this list because I want advise on a CDR loss issue I am
>>> having.
>>>
>>> I want to write the mysql INSERT statement into the SYSLOG or separate
>>> file in case mysql gives any error.
>>>
>>> I have try different ways to accomplish this, by editing db_mysql
>>> module, but I get segfaul, so I guess I am doing something really wrong.
>>>
>>> Can someone give me please some direction here?
>>>
>>> Thanks,
>>>
>>> WC
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
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Re: [OpenSIPS-Users] Write CDR to SYSLOG when mysql statement fail

[Mike Tesliuk]

you can use both , just se the log flag (
http://www.opensips.org/html/docs/modules/1.9.x/acc#id293023 ) and
log_db_flag together (the same for log_missed_flag )


2013/10/11 Wilmar Campos 

> Hi All,
>
> I hope everybody is good!!
> I am writting this list because I want advise on a CDR loss issue I am
> having.
>
> I want to write the mysql INSERT statement into the SYSLOG or separate
> file in case mysql gives any error.
>
> I have try different ways to accomplish this, by editing db_mysql module,
> but I get segfaul, so I guess I am doing something really wrong.
>
> Can someone give me please some direction here?
>
> Thanks,
>
> WC
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
___
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Re: [OpenSIPS-Users] Opensip as transaprent inbound proxy for asterisk

[Mike Tesliuk]

the sugestion from Stefano is to you transport the ip information from
opensips to asterisk, when you are in asterisk you get that variable and
validate the customer, if just opensips will talk with asterisk so you dont
need the ip address on asterisk, just in opensips, you made all validation
on opensips, and send that with some kind of identification of the customer
to asterisk, define as callerid or accountcode and make the next step on
asterisk.

I understand what you want, maybe you should try to use some kind of
redirect, you send the call to opensips, opensips check which server should
receive that and so you reply with a redirect message and the call will be
stablished directly with the asterisk, opensips will just show the path


2013/10/10 bluerain 

> Just FYI, I tried, I insert your line in the method invite and right before
> the routing, Asterisk didn't seem to care.  It still care about the prior
> Hop IP.
>
> So what I mean is that
>
> from 199.33.33.33 --> opensip 22.55.33.33 (and then I put your line) -->
> Asterisk server.
>
> Asterisk server identified the call came from 22.55.33.33 and not
> 199.33.33.33
>
> Frank
>
>
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/Opensip-as-transaprent-inbound-proxy-for-asterisk-tp7588047p7588062.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
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Re: [OpenSIPS-Users] Opensips 1.10 + Freeradius

[Mike Tesliuk]

on the client i think that should be $INCLUDE not INCLUDE


2013/10/7 Rodrigo Ferreira 

> I did Mike.
>
> ### radiusclient dictionary
>
> root@opensips:/etc# cat radiusclient-ng/dictionary | grep opensips
> INCLUDE dictionary.opensips
> root@opensips:/etc# ls radiusclient-ng/dictionary.opensips
> radiusclient-ng/dictionary.opensips
>
>
> ### freeradius dictionary
> root@opensips:/etc# cat freeradius/dictionary | grep opensips
> $INCLUDE/etc/freeradius/dictionary.opensips
> root@opensips:/etc# ls /etc/freeradius/dictionary.opensips
> /etc/freeradius/dictionary.opensips
>
>
>
>
> Atenciosamente.
> Eng.° Rodrigo Ferreira
> ITIL v3 Certified
>
> <http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901>
>
>
> 2013/10/7 Mike Tesliuk 
>
>> Rodrigo,
>>
>> did you check if this file is being included by your main dictionary file
>> ?  ( /etc/radiusclient-ng/dictionary ) (on client and on the server)
>>
>>
>> 2013/10/7 Rodrigo Ferreira 
>>
>>>  Hi guys,
>>>
>>> I'm stucked trying to get my freeradius working with my OpenSIPs. I'm
>>> getting this error from my OpenSIPs.
>>>
>>> Oct  7 12:00:00 opensips /sbin/opensips[11265]: rc_avpair_new: unknown
>>> attribute 0
>>> Oct  7 12:00:00 opensips /sbin/opensips[11265]:
>>> ERROR:aaa_radius:rad_avp_add: failure
>>> Oct  7 12:00:00 opensips /sbin/opensips[11265]:
>>> ERROR:acc:acc_aaa_request: failed to add Sip-Response-Code, 2
>>>
>>> I saw a few topics that tells me to check my radiusclient dictionary,
>>> but I did that several times, and its ok, both freeradius and radiusclient
>>> use the same dictionary.
>>>
>>> root@opensips:/etc/radiusclient-ng# cat
>>> /etc/freeradius/dictionary.opensips
>>> #
>>> # SIP RADIUS attributes
>>> #
>>> # Schulzrinne indicates attributes according to
>>> # draft-schulzrinne-sipping-radius-accounting-00
>>> #
>>> # Proprietary indicates an attribute that hasn't
>>> # been standardized
>>> #
>>> #
>>> # NOTE: All standard (IANA registered) attributes are
>>> #   defined in the default dictionary of the
>>> #   radiusclient-ng library.
>>> #
>>>
>>> ### Take standard attributes from the radiusclient-ng dictionaries ###
>>>
>>> #$INCLUDE   /etc/radiusclient-ng/dictionary
>>>
>>> ### SIP Attributes ###
>>> ATTRIBUTE Sip-Method   101  integer# Schulzrinne, acc
>>> ATTRIBUTE Sip-Response-Code102  integer# Schulzrinne, acc
>>> ATTRIBUTE Sip-Cseq 103  string # Schulzrinne, acc
>>> ATTRIBUTE Sip-To-Tag   104  string # Schulzrinne, acc
>>> ATTRIBUTE Sip-From-Tag 105  string # Schulzrinne, acc
>>> ATTRIBUTE Sip-Branch-ID106  string
>>> ATTRIBUTE Sip-Translated-Request-URI   107  string # Proprietary, acc
>>> ATTRIBUTE Sip-Uri-User 208  string # Proprietary,
>>> auth_radius
>>> ATTRIBUTE Sip-Group211  string # Proprietary,
>>> group_radius
>>> ATTRIBUTE Sip-Rpid 213  string # Proprietary,
>>> auth_radius
>>> ATTRIBUTE Billing-Party218  string
>>> ATTRIBUTE SIP-AVP  225  string # Proprietary,
>>> avp_radius
>>>
>>> ### Sip-Method Values ###
>>> VALUE Sip-Method Undefined  0
>>> VALUE Sip-Method Invite 1
>>> VALUE Sip-Method Cancel 2
>>> VALUE Sip-Method Ack4
>>> VALUE Sip-Method Bye8
>>> VALUE Sip-Method Info   16
>>> VALUE Sip-Method Options32
>>> VALUE Sip-Method Update 64
>>> VALUE Sip-Method Register   128
>>> VALUE Sip-Method Message256
>>> VALUE Sip-Method Subscribe  512
>>> VALUE Sip-Method Notify 1024
>>> VALUE Sip-Method Prack  2048
>>> VALUE Sip-Method Refer  4096
>>> VALUE Sip-Method Publish8192
>>> VALUE Sip-Method Other  16384
>>>
>>> ### Sip-Response-Code Values ###
>>> VALUE Sip-Response-Code  Undefined  0
>>> VALUE Sip-Response-Code  Invite 1
>>> VALUE Sip-Response-Code  Cancel 2
&g

Re: [OpenSIPS-Users] Opensips 1.10 + Freeradius

[Mike Tesliuk]

Rodrigo,

did you check if this file is being included by your main dictionary file
?  ( /etc/radiusclient-ng/dictionary ) (on client and on the server)


2013/10/7 Rodrigo Ferreira 

> Hi guys,
>
> I'm stucked trying to get my freeradius working with my OpenSIPs. I'm
> getting this error from my OpenSIPs.
>
> Oct  7 12:00:00 opensips /sbin/opensips[11265]: rc_avpair_new: unknown
> attribute 0
> Oct  7 12:00:00 opensips /sbin/opensips[11265]:
> ERROR:aaa_radius:rad_avp_add: failure
> Oct  7 12:00:00 opensips /sbin/opensips[11265]: ERROR:acc:acc_aaa_request:
> failed to add Sip-Response-Code, 2
>
> I saw a few topics that tells me to check my radiusclient dictionary, but
> I did that several times, and its ok, both freeradius and radiusclient use
> the same dictionary.
>
> root@opensips:/etc/radiusclient-ng# cat
> /etc/freeradius/dictionary.opensips
> #
> # SIP RADIUS attributes
> #
> # Schulzrinne indicates attributes according to
> # draft-schulzrinne-sipping-radius-accounting-00
> #
> # Proprietary indicates an attribute that hasn't
> # been standardized
> #
> #
> # NOTE: All standard (IANA registered) attributes are
> #   defined in the default dictionary of the
> #   radiusclient-ng library.
> #
>
> ### Take standard attributes from the radiusclient-ng dictionaries ###
>
> #$INCLUDE   /etc/radiusclient-ng/dictionary
>
> ### SIP Attributes ###
> ATTRIBUTE Sip-Method   101  integer# Schulzrinne, acc
> ATTRIBUTE Sip-Response-Code102  integer# Schulzrinne, acc
> ATTRIBUTE Sip-Cseq 103  string # Schulzrinne, acc
> ATTRIBUTE Sip-To-Tag   104  string # Schulzrinne, acc
> ATTRIBUTE Sip-From-Tag 105  string # Schulzrinne, acc
> ATTRIBUTE Sip-Branch-ID106  string
> ATTRIBUTE Sip-Translated-Request-URI   107  string # Proprietary, acc
> ATTRIBUTE Sip-Uri-User 208  string # Proprietary,
> auth_radius
> ATTRIBUTE Sip-Group211  string # Proprietary,
> group_radius
> ATTRIBUTE Sip-Rpid 213  string # Proprietary,
> auth_radius
> ATTRIBUTE Billing-Party218  string
> ATTRIBUTE SIP-AVP  225  string # Proprietary,
> avp_radius
>
> ### Sip-Method Values ###
> VALUE Sip-Method Undefined  0
> VALUE Sip-Method Invite 1
> VALUE Sip-Method Cancel 2
> VALUE Sip-Method Ack4
> VALUE Sip-Method Bye8
> VALUE Sip-Method Info   16
> VALUE Sip-Method Options32
> VALUE Sip-Method Update 64
> VALUE Sip-Method Register   128
> VALUE Sip-Method Message256
> VALUE Sip-Method Subscribe  512
> VALUE Sip-Method Notify 1024
> VALUE Sip-Method Prack  2048
> VALUE Sip-Method Refer  4096
> VALUE Sip-Method Publish8192
> VALUE Sip-Method Other  16384
>
> ### Sip-Response-Code Values ###
> VALUE Sip-Response-Code  Undefined  0
> VALUE Sip-Response-Code  Invite 1
> VALUE Sip-Response-Code  Cancel 2
> VALUE Sip-Response-Code  Ack4
> VALUE Sip-Response-Code  Bye8
> VALUE Sip-Response-Code  Info   16
> VALUE Sip-Response-Code  Options32
> VALUE Sip-Response-Code  Update 64
> VALUE Sip-Response-Code  Register   128
> VALUE Sip-Response-Code  Message256
> VALUE Sip-Response-Code  Subscribe  512
> VALUE Sip-Response-Code  Notify 1024
> VALUE Sip-Response-Code  Prack  2048
> VALUE Sip-Response-Code  Refer  4096
> VALUE Sip-Response-Code  Publish8192
> VALUE Sip-Response-Code  Other  16384
>
> ### Acct-Status-Type Values ###
> VALUE Acct-Status-Type Start 1 # RFC2866, acc
> VALUE Acct-Status-Type Stop  2 # RFC2866, acc
> VALUE Acct-Status-Type Failed   15 # RFC2866, acc
>
> ### Service-Type Values ###
> VALUE Service-Type Call-Check   10 # RFC2865,
> uri_radius
> VALUE Service-Type Group-Check  12 # Proprietary,
> group_radius
> VALUE Service-Type Sip-Session  15 # Schulzrinne, acc,
> auth_radius
> VALUE Service-Type SIP-Caller-AVPs  30 # Proprietary,
> avp_radius
> VALUE Service-Type SIP-Callee-AVPs  31 # Proprietary,
> avp_radius
>
> ### Attributes added by AG Projects ###
> ATTRIBUTE   Source-IP  214 string
> ATTRIBUTE   Source-Port215 string
> ATTRIBUTE   Canonical-URI  216 string
> ATTRIBUTE   Delay-Time 217 string
> ATTRIBUTE   Divert-Reason  219 string
> ATTRIBUTE   X-RTP-Stat 220 string
> ATTRIBUTE   From-Header221 string
> ATTRIBUTE   User-Agent  

Re: [OpenSIPS-Users] Opensips 1.10 NAT

[Mike Tesliuk]

probably if the UA's are on the same network, when you send the package the
package is going with the external ip on the SDP , when the call is
stablished probably you router is not allowing to open the second lag
because the UA's are trying to stablish from inside using the outside ip
addressl, so when you go through rtpproxy this not happen  both sides use
the opensips (rtpproxy) ip address to sdp.

If my logic is not correct please somebody let me know.


2013/10/4 Rodrigo Ferreira 

> Forcing the traffic through RTPPROXY worked, but why isnt working the
> nat_uac_test?
>
> Kinda weird
>
>
>
> Atenciosamente.
> Eng.° Rodrigo Ferreira
> ITIL v3 Certified
>
> <http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901>
>
>
> 2013/10/4 Mike Tesliuk 
>
>> well, probably you softphone/ip phone, is using some kind of stun or
>> other kind of nat features, so, nothing come to be detected, this can
>> happen, so, if you will be ever using nat, you can force the rtpproxy
>> without nat detection, this will solve your problem, if you read the
>> documentation (
>> http://www.opensips.org/html/docs/modules/1.9.x/nathelper.html#id292854) you 
>> can see that this test are made over rfc1918 or different ip address
>> from via and signalling
>>
>> The problem is probably the fact that when the call is stablished, the
>> media cannot traverse, you have the correct ip information on sdp but the
>> router does not permit the session to be opened, so, do a test forcing the
>> use of rtpproxy without the nat detection, just force all trafic throught
>> rtpproxy
>>
>>
>> 2013/10/4 Rodrigo Ferreira 
>>
>>> I did that Mike ..
>>>
>>> my "nat_uac_client" isnt passing in any verification ...
>>>
>>> I did this ..
>>>
>>> if ( nat_uac_test("1") ) xlog("UAC TEST = 1");
>>>
>>> if ( nat_uac_test("2") ) xlog("UAC TEST = 2");
>>>
>>> if ( nat_uac_test("4") ) xlog("UAC TEST = 4");
>>>
>>> if ( nat_uac_test("8") ) xlog("UAC TEST = 8");
>>>
>>> if ( nat_uac_test("16") ) xlog("UAC TEST = 16");
>>>
>>> if ( nat_uac_test("32") ) xlog("UAC TEST = 32");
>>>
>>> if ( nat_uac_test("64") ) xlog("UAC TEST = 64");
>>>
>>> in the beginning of the script, to see what is happening to my NAT, and
>>> i got nothing.
>>>
>>>
>>>
>>> Atenciosamente.
>>> Eng.° Rodrigo Ferreira
>>> ITIL v3 Certified
>>>
>>> <http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901>
>>>
>>>
>>> 2013/10/4 Mike Tesliuk 
>>>
>>>> That howto is just a sample (with a lot of comments) to better
>>>> understand of nat configuration (over my understand offcourse), so, you can
>>>> check and compare with your configuration to identify about something
>>>> missing
>>>>
>>>>
>>>>
>>>>
>>>> 2013/10/4 Rodrigo Ferreira 
>>>>
>>>>> Yes I did Mike,
>>>>>
>>>>> and my SIP messages are ok.
>>>>>
>>>>> I will take a look at your tutorial.
>>>>>
>>>>> tks
>>>>>
>>>>>
>>>>>
>>>>> Atenciosamente.
>>>>> Eng.° Rodrigo Ferreira
>>>>> ITIL v3 Certified
>>>>>
>>>>> <http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901>
>>>>>
>>>>>
>>>>> 2013/10/3 Mike Tesliuk 
>>>>>
>>>>>> Did you try to made some debug rodrigo ? maybe some rule is missing
>>>>>> on your route script
>>>>>>
>>>>>> i made a tutorial over version 1.9 that you can check
>>>>>>
>>>>>> [portugues]
>>>>>> http://opensips.com.br/wiki/index.php?title=Opensips_NAT_Script_com_RTPproxy
>>>>>> [english]
>>>>>> http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_English
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> 2013/10/3 Rodrigo Ferreira 
>>>>>>
>>>>>>>  Hi guys,
>>>>>>

Re: [OpenSIPS-Users] Opensips 1.10 NAT

[Mike Tesliuk]

well, probably you softphone/ip phone, is using some kind of stun or other
kind of nat features, so, nothing come to be detected, this can happen, so,
if you will be ever using nat, you can force the rtpproxy without nat
detection, this will solve your problem, if you read the documentation (
http://www.opensips.org/html/docs/modules/1.9.x/nathelper.html#id292854 )
you can see that this test are made over rfc1918 or different ip address
from via and signalling

The problem is probably the fact that when the call is stablished, the
media cannot traverse, you have the correct ip information on sdp but the
router does not permit the session to be opened, so, do a test forcing the
use of rtpproxy without the nat detection, just force all trafic throught
rtpproxy


2013/10/4 Rodrigo Ferreira 

> I did that Mike ..
>
> my "nat_uac_client" isnt passing in any verification ...
>
> I did this ..
>
> if ( nat_uac_test("1") ) xlog("UAC TEST = 1");
>
> if ( nat_uac_test("2") ) xlog("UAC TEST = 2");
>
> if ( nat_uac_test("4") ) xlog("UAC TEST = 4");
>
> if ( nat_uac_test("8") ) xlog("UAC TEST = 8");
>
> if ( nat_uac_test("16") ) xlog("UAC TEST = 16");
>
> if ( nat_uac_test("32") ) xlog("UAC TEST = 32");
>
> if ( nat_uac_test("64") ) xlog("UAC TEST = 64");
>
> in the beginning of the script, to see what is happening to my NAT, and i
> got nothing.
>
>
>
> Atenciosamente.
> Eng.° Rodrigo Ferreira
> ITIL v3 Certified
>
> <http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901>
>
>
> 2013/10/4 Mike Tesliuk 
>
>> That howto is just a sample (with a lot of comments) to better understand
>> of nat configuration (over my understand offcourse), so, you can check and
>> compare with your configuration to identify about something missing
>>
>>
>>
>>
>> 2013/10/4 Rodrigo Ferreira 
>>
>>> Yes I did Mike,
>>>
>>> and my SIP messages are ok.
>>>
>>> I will take a look at your tutorial.
>>>
>>> tks
>>>
>>>
>>>
>>> Atenciosamente.
>>> Eng.° Rodrigo Ferreira
>>> ITIL v3 Certified
>>>
>>> <http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901>
>>>
>>>
>>> 2013/10/3 Mike Tesliuk 
>>>
>>>> Did you try to made some debug rodrigo ? maybe some rule is missing on
>>>> your route script
>>>>
>>>> i made a tutorial over version 1.9 that you can check
>>>>
>>>> [portugues]
>>>> http://opensips.com.br/wiki/index.php?title=Opensips_NAT_Script_com_RTPproxy
>>>> [english]
>>>> http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_English
>>>>
>>>>
>>>>
>>>>
>>>> 2013/10/3 Rodrigo Ferreira 
>>>>
>>>>>  Hi guys,
>>>>>
>>>>> After a long time without using Opensips (almost a year) I tried to
>>>>> install the opensips 1.10 and everything went well BUT when I make a call,
>>>>> there's no audio, I know that is something because of NAT, but I have the
>>>>> nathelper and rtpproxy configuration on my opensips.cfg.
>>>>>
>>>>> There's anything else that I could take a look at?
>>>>>
>>>>> Thanks
>>>>>
>>>>>
>>>>> Atenciosamente.
>>>>> Eng.° Rodrigo Ferreira
>>>>>  ITIL v3 Certified
>>>>>
>>>>> <http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901>
>>>>>
>>>>> ___
>>>>> Users mailing list
>>>>> Users@lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>
>>>> ___
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>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>
>>> ___
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>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
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>>
>>
>
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Re: [OpenSIPS-Users] Opensips 1.10 NAT

[Mike Tesliuk]

That howto is just a sample (with a lot of comments) to better understand
of nat configuration (over my understand offcourse), so, you can check and
compare with your configuration to identify about something missing




2013/10/4 Rodrigo Ferreira 

> Yes I did Mike,
>
> and my SIP messages are ok.
>
> I will take a look at your tutorial.
>
> tks
>
>
>
> Atenciosamente.
> Eng.° Rodrigo Ferreira
> ITIL v3 Certified
>
> <http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901>
>
>
> 2013/10/3 Mike Tesliuk 
>
>> Did you try to made some debug rodrigo ? maybe some rule is missing on
>> your route script
>>
>> i made a tutorial over version 1.9 that you can check
>>
>> [portugues]
>> http://opensips.com.br/wiki/index.php?title=Opensips_NAT_Script_com_RTPproxy
>> [english]
>> http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_English
>>
>>
>>
>>
>> 2013/10/3 Rodrigo Ferreira 
>>
>>>  Hi guys,
>>>
>>> After a long time without using Opensips (almost a year) I tried to
>>> install the opensips 1.10 and everything went well BUT when I make a call,
>>> there's no audio, I know that is something because of NAT, but I have the
>>> nathelper and rtpproxy configuration on my opensips.cfg.
>>>
>>> There's anything else that I could take a look at?
>>>
>>> Thanks
>>>
>>>
>>> Atenciosamente.
>>> Eng.° Rodrigo Ferreira
>>>  ITIL v3 Certified
>>>
>>> <http://br.linkedin.com/pub/rodrigo-ferreira/31/757/901>
>>>
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>> ___
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>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
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Re: [OpenSIPS-Users] Opensips 1.10 NAT

[Mike Tesliuk]

Did you try to made some debug rodrigo ? maybe some rule is missing on your
route script

i made a tutorial over version 1.9 that you can check

[portugues]
http://opensips.com.br/wiki/index.php?title=Opensips_NAT_Script_com_RTPproxy
[english]
http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_English




2013/10/3 Rodrigo Ferreira 

> Hi guys,
>
> After a long time without using Opensips (almost a year) I tried to
> install the opensips 1.10 and everything went well BUT when I make a call,
> there's no audio, I know that is something because of NAT, but I have the
> nathelper and rtpproxy configuration on my opensips.cfg.
>
> There's anything else that I could take a look at?
>
> Thanks
>
>
> Atenciosamente.
> Eng.° Rodrigo Ferreira
> ITIL v3 Certified
>
> 
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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Re: [OpenSIPS-Users] FW: Audio and Video not working for different otuside netrwork

[Mike Tesliuk]

I send a link on last message but the moderation block because of other
informations that i sent

this is a simple cfg file with comments trying to show how to implement the
nat, off course this will execute nat functions for any connection but is
just try to explain how this work (if i understand correct) :)

http://opensips.com.br/wiki/index.php?title=Oopensips_Nat_script_with_RTPPROXY_-_English


if somebody see any error, please tellme and i will fix.

Thanks



2013/9/29 kamika 

> If one of your clients or both are behind NAT, you will certanly have
> issues
> with voice. You have to implement STUN or/and RTPProxy or Mediaproxy with
> Nathelper module depending on how you want to serve you clients. Opensips
> is
> not the one that may be used right out of the box. You have to learn both
> sip specifications and opensips pseudo language. Otherwise you won't be
> sure
> your opensips is well protected and works as you want.
>
>
>
>
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/FW-Audio-and-Video-not-working-for-different-otuside-netrwork-tp7587910p7587945.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
> ___
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Re: [OpenSIPS-Users] FW: Audio and Video not working for different otuside netrwork

[Mike Tesliuk]

If possible, paste your route file too


2013/9/27 Mike Tesliuk 

> start your opensips in debug mode, try to make the call, get all the
> message and paste in some pastebin website and show us the link
>
>
>
> 2013/9/27 Rajesh Babu 
>
>> I am getting Error 483, too many Hops, There is no other error messages i
>> am getting. Please some one help me out in this
>>
>> ** **
>>
>> *From:* users-boun...@lists.opensips.org [mailto:
>> users-boun...@lists.opensips.org] *On Behalf Of *Rajesh Babu
>> *Sent:* Friday, 27 September, 2013 6:08 PM
>>
>> *To:* 'OpenSIPS users mailling list'
>> *Subject:* Re: [OpenSIPS-Users] FW: Audio and Video not working for
>> different otuside netrwork
>>
>> ** **
>>
>> HI Mike,
>>
>> ** **
>>
>>Now the RTP is up and i am getting this message on my logs
>>
>> [root@centos64 rtpproxy-1.2.0]# tailf /var/log/messages
>>
>> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17800]:
>> INFO:core:probe_max_sock_buff: using rcv buffer of 448 kb
>>
>> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17810]:
>> INFO:rtpproxy:rtpp_test: rtp proxy  found,
>> support for it enabled
>>
>> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17812]:
>> INFO:rtpproxy:rtpp_test: rtp proxy  found,
>> support for it enabled
>>
>> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17815]:
>> INFO:rtpproxy:rtpp_test: rtp proxy  found,
>> support for it enabled
>>
>> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17808]:
>> INFO:rtpproxy:rtpp_test: rtp proxy  found,
>> support for it enabled
>>
>> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17811]:
>> INFO:rtpproxy:rtpp_test: rtp proxy  found,
>> support for it enabled
>>
>> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17809]:
>> INFO:rtpproxy:rtpp_test: rtp proxy  found,
>> support for it enabled
>>
>> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17814]:
>> INFO:rtpproxy:rtpp_test: rtp proxy  found,
>> support for it enabled
>>
>> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17809]:
>> WARNING:drouting:dr_load_routing_info: table "dr_rules" is empty
>>
>> Sep 28 01:59:43 centos64 opensips: INFO:core:daemonize: pre-daemon
>> process exiting with 0
>>
>> ** **
>>
>> But my test tool is not connecting back my server. Is there any mistake i
>> am doing.
>>
>> ** **
>>
>> Thanks 
>>
>> Rajesh
>>
>> ** **
>>
>> *From:* users-boun...@lists.opensips.org [
>> mailto:users-boun...@lists.opensips.org]
>> *On Behalf Of *Rajesh Babu
>> *Sent:* Friday, 27 September, 2013 2:34 PM
>>
>> *To:* 'OpenSIPS users mailling list'
>> *Subject:* Re: [OpenSIPS-Users] FW: Audio and Video not working for
>> different otuside netrwork
>>
>> ** **
>>
>> Hi Mike,
>>
>> ** **
>>
>>   This is log i am geting wheni try to start the service
>>
>> ** **
>>
>> [root@centos64 rtpproxy-1.2.0]# tailf /var/log/messages
>>
>> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12839]:
>> WARNING:rtpproxy:rtpp_test: support for RTP proxy  has
>> been disabled temporarily
>>
>> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]:
>> ERROR:rtpproxy:send_rtpp_command: can't send command to a RTP proxy
>> Connection refused
>>
>> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]:
>> ERROR:rtpproxy:send_rtpp_command: proxy  does not
>> respond, disable it
>>
>> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]:
>> WARNING:rtpproxy:rtpp_test: can't get version of the RTP proxy
>>
>> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]:
>> WARNING:rtpproxy:rtpp_test: support for RTP proxy  has
>> been disabled temporarily
>>
>> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]:
>> ERROR:rtpproxy:send_rtpp_command: can't send command to a RTP proxy
>> Connection refused
>>
>> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]:
>> ERROR:rtpproxy:send_rtpp_command: proxy  does not
>> respond, disable it
>>
>> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]:

Re: [OpenSIPS-Users] FW: Audio and Video not working for different otuside netrwork

[Mike Tesliuk]

start your opensips in debug mode, try to make the call, get all the
message and paste in some pastebin website and show us the link



2013/9/27 Rajesh Babu 

> I am getting Error 483, too many Hops, There is no other error messages i
> am getting. Please some one help me out in this
>
> ** **
>
> *From:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *On Behalf Of *Rajesh Babu
> *Sent:* Friday, 27 September, 2013 6:08 PM
>
> *To:* 'OpenSIPS users mailling list'
> *Subject:* Re: [OpenSIPS-Users] FW: Audio and Video not working for
> different otuside netrwork
>
> ** **
>
> HI Mike,
>
> ** **
>
>Now the RTP is up and i am getting this message on my logs
>
> [root@centos64 rtpproxy-1.2.0]# tailf /var/log/messages
>
> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17800]:
> INFO:core:probe_max_sock_buff: using rcv buffer of 448 kb
>
> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17810]:
> INFO:rtpproxy:rtpp_test: rtp proxy  found, support
> for it enabled
>
> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17812]:
> INFO:rtpproxy:rtpp_test: rtp proxy  found, support
> for it enabled
>
> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17815]:
> INFO:rtpproxy:rtpp_test: rtp proxy  found, support
> for it enabled
>
> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17808]:
> INFO:rtpproxy:rtpp_test: rtp proxy  found, support
> for it enabled
>
> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17811]:
> INFO:rtpproxy:rtpp_test: rtp proxy  found, support
> for it enabled
>
> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17809]:
> INFO:rtpproxy:rtpp_test: rtp proxy  found, support
> for it enabled
>
> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17814]:
> INFO:rtpproxy:rtpp_test: rtp proxy  found, support
> for it enabled
>
> Sep 28 01:59:43 centos64 /usr/local/server/pbx/sbin/opensips[17809]:
> WARNING:drouting:dr_load_routing_info: table "dr_rules" is empty
>
> Sep 28 01:59:43 centos64 opensips: INFO:core:daemonize: pre-daemon process
> exiting with 0
>
> ** **
>
> But my test tool is not connecting back my server. Is there any mistake i
> am doing.
>
> ** **
>
> Thanks 
>
> Rajesh
>
> ** **
>
> *From:* users-boun...@lists.opensips.org [
> mailto:users-boun...@lists.opensips.org ]
> *On Behalf Of *Rajesh Babu
> *Sent:* Friday, 27 September, 2013 2:34 PM
>
> *To:* 'OpenSIPS users mailling list'
> *Subject:* Re: [OpenSIPS-Users] FW: Audio and Video not working for
> different otuside netrwork
>
> ** **
>
> Hi Mike,
>
> ** **
>
>   This is log i am geting wheni try to start the service
>
> ** **
>
> [root@centos64 rtpproxy-1.2.0]# tailf /var/log/messages
>
> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12839]:
> WARNING:rtpproxy:rtpp_test: support for RTP proxy  has
> been disabled temporarily
>
> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]:
> ERROR:rtpproxy:send_rtpp_command: can't send command to a RTP proxy
> Connection refused
>
> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]:
> ERROR:rtpproxy:send_rtpp_command: proxy  does not
> respond, disable it
>
> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]:
> WARNING:rtpproxy:rtpp_test: can't get version of the RTP proxy
>
> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12833]:
> WARNING:rtpproxy:rtpp_test: support for RTP proxy  has
> been disabled temporarily
>
> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]:
> ERROR:rtpproxy:send_rtpp_command: can't send command to a RTP proxy
> Connection refused
>
> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]:
> ERROR:rtpproxy:send_rtpp_command: proxy  does not
> respond, disable it
>
> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]:
> WARNING:rtpproxy:rtpp_test: can't get version of the RTP proxy
>
> Sep 27 22:29:03 centos64 /usr/local/server/pbx/sbin/opensips[12832]:
> WARNING:rtpproxy:rtpp_test: support for RTP proxy  has
> been disabled temporarily
>
> Sep 27 22:29:03 centos64 opensips: INFO:core:daemonize: pre-daemon process
> exiting with 0
>
> ** **
>
> *From:* users-boun...@lists.opensips.org [
> mailto:users-boun...@lists.opensips.org ]
> *On Behalf Of *Mike Tesliuk
> *Sent:* Thursday, 26 September, 2013 10:25 PM
>
> *To:* OpenS

Re: [OpenSIPS-Users] FW: Audio and Video not working for different otuside netrwork

[Mike Tesliuk]

When you use the residential script almost all configuration come alredy
working for this

i have a tutorial (in portuguese ( i think that i should translate to
english :))) , where you can see a routing script working with nat

http://opensips.com.br/wiki/index.php?title=Opensips_1.9


You can take a look at modules documentation too

http://www.opensips.org/html/docs/modules/1.9.x/nathelper.html
http://www.opensips.org/html/docs/modules/1.9.x/rtpproxy.html

There is on this maillist too a lot of discussions about this, below you
can see one case

http://opensips.org/pipermail/users/2011-January/016130.html

If you get some information from an old version of opensips probably will
be necessary to take a look on the module documentation to check about
little diferences , but i think that this is the start point :)

and if you is new to opensips i recommend to you the book about opensips (
http://www.packtpub.com/building-telephony-systems-with-opensips-1-6/book )






2013/9/26 Rajesh Babu 

> Hi Mike,
>
> ** **
>
>   Thanks for the response, I am totally new to this world, can you please
> help me by directing to on how to configure links. It will be great. 
>
> Thanks in advance
>
> Regards
>
> Rajesh
>
> ** **
>
> *From:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *On Behalf Of *Mike Tesliuk
> *Sent:* Thursday, 26 September, 2013 12:25 PM
> *To:* OpenSIPS users mailling list
> *Subject:* Re: [OpenSIPS-Users] FW: Audio and Video not working for
> different otuside netrwork
>
> ** **
>
> you should configure the nathelper and rtpproxy, this should help in you
> issue.
>
> ** **
>
> 2013/9/26 Rajesh Babu 
>
> Hi,
>
>  
>
>I am new to the OpenSIP world. I have installed a OpenSIP on my
> network. If i make a Call inside the network between two users i don’t have
> any issue, where as from outside the network, even though i can see the
> user registered in my server i am not able to call registered user (I see
> the user in my UL show listing). The call is established but i am not able
> to talk (Mean the audio and video are not getting transffered).
>
>  
>
> Where as messages are going fine without any issue. I guess it is because
> message transmit over XMPP where calls on SIP right.
>
>  
>
>  
>
> I am really struck and i don’t know how to proceed, please help me out
>
>  
>
>  
>
>  
>
> Thanks
>
> Rajesh
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> ** **
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
___
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Re: [OpenSIPS-Users] FW: Audio and Video not working for different otuside netrwork

[Mike Tesliuk]

you should configure the nathelper and rtpproxy, this should help in you
issue.


2013/9/26 Rajesh Babu 

> Hi,
>
> ** **
>
>I am new to the OpenSIP world. I have installed a OpenSIP on my
> network. If i make a Call inside the network between two users i don’t have
> any issue, where as from outside the network, even though i can see the
> user registered in my server i am not able to call registered user (I see
> the user in my UL show listing). The call is established but i am not able
> to talk (Mean the audio and video are not getting transffered).
>
> ** **
>
> Where as messages are going fine without any issue. I guess it is because
> message transmit over XMPP where calls on SIP right.
>
> ** **
>
> ** **
>
> I am really struck and i don’t know how to proceed, please help me out
>
> ** **
>
> ** **
>
> ** **
>
> Thanks
>
> Rajesh
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
___
Users mailing list
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