14 Nov
2018
14 Nov
'18
12:39 p.m.
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 $ru (rewriteuri, rewriteuser,
rewriteuserpass?)
Thank you,
Atenciosamente,
2018-11-14 7:43 GMT-02:00 David Villasmil <david.villasmil.work(a)gmail.com>om>:
You would be authenticating the calls TWICE, is this
what you want?
I found the easiest would be to just configure in your asterisk the
Kamailio gateway as the proxy.
I haven’t worked with asterisk in a while, but in freeswitch al that’s
needed is adding a parameter called “proxy” and set the Kamailio’s up
address. All this with Kamailio’s default configuration.
This would only work if both asterisk and Kamailio have public addresses.
No RTP proxying, they’d go straight to the provider.
Regards,
David
On Wed, 14 Nov 2018 at 02:07, Valter Nogueira <valter(a)fastway.com.br>
wrote:
Hi Henning,
thanks for your tip.
I just checked it and I am sure it will be valuable.
Atenciosamente,
2018-11-13 19:04 GMT-02:00 Henning Westerholt <hw(a)kamailio.org>rg>:
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Am Freitag, 9. November 2018, 21:25:15 CET
schrieb Valter Nogueira:
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Today, I use Asterisk as a SIP/RTP PROXY
I proxy from customers Asterisks to a VOIP provider, in a multi-homed
server.
Now, I want to move to Kamailio without any rupture in customer's
configuration.
As anyone can imagine I am kind of lost.
USER ACCOUNTS
In Asterisk, I create a dynamic host account named ACCOUNT1 and I
receive
in *FROM HEADER
sip:ACCOUNT1@customer_ip_address*
In Kamailio, I have to define the account's domain like *kamctl add
ACCOUNT1(a)mydomain.com <ACCOUNT1(a)mydomain.com> password. *Kamailio just
accepts a REGISTER/INVITE from *ACCOUNT1(a)mydomain.com
<ACCOUNT1(a)mydomain.com>*
SIP/RTP PROXY
In Asterisk, I just dialout to the VOIP PROVIDER like *dial
(SIP/VOIP_ACCOUNT/${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