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

[Bruce Wang]

Hi bluerain,

Our production setup is using OpenSIPS as proxy to many Asterisk instances
via load_balancer module.

I can see from the SIP logs on our Asterisk servers the SIP headers sent
from OpenSIPS to Asterisk still contains the original IP (From, Contact,
and the bottom one Via)

Have you tried to read any of those SIP header in your Asterisk script?

You should also check your opensips script in /etc/opensips/opensips.cfg to
see if you have rewrite SIP headers which contain the original ip.

Cheers,
Bruce


On Fri, Oct 11, 2013 at 11:38 AM, Mike Tesliuk <mike at ultra.net.br> wrote:

> 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 <frank21118 at yahoo.com>
>
>> 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
>>
>>
>>
>> --
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>> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>>
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-- 
simple is good
http://brucewang.net
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