[OpenSIPS-Users] Opensips 1.10 NAT

Rodrigo Ferreira rsferreira08 at gmail.com
Fri Oct 4 16:50:55 CEST 2013


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 <mike at ultra.net.br>

> 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 <rsferreira08 at gmail.com>
>
>> 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 <mike at ultra.net.br>
>>
>>> 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 <rsferreira08 at gmail.com>
>>>
>>>>  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>
>>>>
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>>>>
>>>>
>>>
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>>
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>
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