[SR-Users] from Asterisk to Kamailio

[David Villasmil]

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

ᐧ

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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>>>
>>
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>>
>>
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