[SR-Users] from Asterisk to Kamailio

Daniel-Constantin Mierla miconda at gmail.com
Tue Nov 20 17:42:31 CET 2018


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#dispatcher.ex.config

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 at gmail.com
> <mailto:david.villasmil.work at gmail.com>
> phone: +34669448337
>
>>
> On Wed, Nov 14, 2018 at 11:43 AM Valter Nogueira
> <valter at fastway.com.br <mailto:valter at 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 at gmail.com
>     <mailto: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 <mailto: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 <mailto: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 <mailto:ACCOUNT1 at mydomain.com>
>                 <ACCOUNT1 at mydomain.com <mailto:ACCOUNT1 at mydomain.com>>
>                 password. *Kamailio just
>                 > accepts a REGISTER/INVITE from
>                 *ACCOUNT1 at mydomain.com <mailto:ACCOUNT1 at mydomain.com>
>                 > <ACCOUNT1 at mydomain.com <mailto: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
>
>
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>
>
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>
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-- 
Daniel-Constantin Mierla -- www.asipto.com
www.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

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