[SR-Users] from Asterisk to Kamailio

David Villasmil david.villasmil.work at gmail.com
Wed Nov 21 01:40:22 CET 2018


Oh, sorry. I always add it and has come to be default on my config ;)

Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337

ᐧ

On Tue, Nov 20, 2018 at 4:45 PM Daniel-Constantin Mierla <miconda at 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#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
> phone: +34669448337
>
>>
> On Wed, Nov 14, 2018 at 11:43 AM Valter Nogueira <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
>> >:
>>
>>> 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
>>>>>
>>>>
>>>> _______________________________________________
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>>>> sr-users at lists.kamailio.org
>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>
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>>>
>>>
>> _______________________________________________
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>>
>
> _______________________________________________
> Kamailio (SER) - Users Mailing Listsr-users at 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
>
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