20 Nov
2018
20 Nov
'18
7:40 p.m.
Oh, sorry. I always add it and has come to be default on my config ;)
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
ᐧ
On Tue, Nov 20, 2018 at 4:45 PM Daniel-Constantin Mierla <miconda(a)gmail.com>
wrote:
A small correction: there is no WITH_DISPATCHER in the
default switch, but
the module readme has a rather complete example:
-
https://www.kamailio.org/docs/modules/stable/modules/dispatcher.html#dispat…
Cheers,
Daniel
On 17.11.18 01:02, David Villasmil wrote:
Your best bet is to use the default config, and enable:
#!define WITH_MYSQL
#!define WITH_AUTH
#!define WITH_DISPATCHER
add your users to the db, and the gateways to the dispatcher table with
setid 1, you'll be good to go
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
ᐧ
On Wed, Nov 14, 2018 at 11:43 AM Valter Nogueira <valter(a)fastway.com.br>
wrote:
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
_______________________________________________
Kamailio (SER) - Users Mailing
Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda --
www.linkedin.com/in/miconda
Kamailio World Conference -- www.kamailioworld.com
Kamailio Advanced Training, Nov 12-14, 2018, in Berlin -- www.asipto.com
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
:
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:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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>:
> 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@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
>
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users