[SR-Users] from Asterisk to Kamailio

[Valter Nogueira]

I am going to replace the asterisk.

Currenty, asterisk proxies all call incoming into a public ip address to a
third party gateway in a non-public ip address.

Asterisk server has 2 NICs.

Asterisk authenticates incoming call and autheticates itself in sip gateway.

I just need to forward the call and proxy RTP

I guess that is a matter of rewriting $ru (rewriteuri, rewriteuser,
rewriteuserpass?)

Thank you,



Atenciosamente,



2018-11-14 7:43 GMT-02:00 David Villasmil <david.villasmil.work at gmail.com>:

> You would be authenticating the calls TWICE, is this what you want?
> I found the easiest would be to just configure in your asterisk the
> Kamailio gateway as the proxy.
>
> I haven’t worked with asterisk in a while, but in freeswitch al that’s
> needed is adding a parameter called “proxy” and set the Kamailio’s up
> address. All this with Kamailio’s default configuration.
>
> This would only work if both asterisk and Kamailio have public addresses.
> No RTP proxying, they’d go straight to the provider.
>
> Regards,
>
> David
> On Wed, 14 Nov 2018 at 02:07, Valter Nogueira <valter at fastway.com.br>
> wrote:
>
>> Hi Henning,
>>
>> thanks for your tip.
>>
>> I just checked it and I am sure it will be valuable.
>>
>>
>>
>>
>>
>> Atenciosamente,
>>
>>
>>
>> 2018-11-13 19:04 GMT-02:00 Henning Westerholt <hw at kamailio.org>:
>>
>>> Am Freitag, 9. November 2018, 21:25:15 CET schrieb Valter Nogueira:
>>> > Today, I use Asterisk as a SIP/RTP PROXY
>>> >
>>> > I proxy from customers Asterisks to a VOIP provider, in a multi-homed
>>> > server.
>>> >
>>> > Now, I want to move to Kamailio without any rupture in customer's
>>> > configuration.
>>> >
>>> > As anyone can imagine I am kind of lost.
>>> >
>>> > USER ACCOUNTS
>>> >
>>> > In Asterisk, I create a dynamic host account named ACCOUNT1 and I
>>> receive
>>> > in *FROM HEADER sip:ACCOUNT1 at customer_ip_address*
>>> >
>>> > In Kamailio, I have to define the account's domain like *kamctl add
>>> > ACCOUNT1 at mydomain.com <ACCOUNT1 at mydomain.com> password. *Kamailio just
>>> > accepts a REGISTER/INVITE from *ACCOUNT1 at mydomain.com
>>> > <ACCOUNT1 at mydomain.com>*
>>> >
>>> >
>>> > SIP/RTP PROXY
>>> >
>>> > In Asterisk, I just dialout to the VOIP PROVIDER like *dial
>>> > (SIP/VOIP_ACCOUNT/${EXTENSION})*
>>> >
>>> > Asterisk does all the magic (it is a B2BUA). It bridges the new call
>>> and
>>> > media to the original call. Moreover, user don't know anything about
>>> how
>>> > call are completed, nor how credentials are setup and soon.
>>> >
>>> > In Kamailio, I guess that I have to use nat, tm and rtpproxy modules
>>> and
>>> > maybe uac. I am not sure how to setup it.
>>> >
>>> > Can someone send me a clue?
>>>
>>> Hello Valter,
>>>
>>> did you already looked into this tutorials? They are for a bit older
>>> version
>>> of Kamailio and asterisk, but should give you ideas about the direction.
>>>
>>> https://kb.asipto.com/asterisk:index
>>>
>>> Best regards,
>>>
>>> Henning
>>>
>>> --
>>> Henning Westerholt - https://skalatan.de/blog/
>>> Kamailio services - https://skalatan.de/services
>>> Kamailio security assessment - https://skalatan.de/de/assessment
>>>
>>
>> _______________________________________________
>> Kamailio (SER) - Users Mailing List
>> sr-users at lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users at lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20181114/31d42e3d/attachment.html>