9 Nov
2018
9 Nov
'18
11:25 p.m.
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/${EXTENSION})*
Asterisk does all the magic (it is a B2BUA). It bridges the new call and
media to the original call. Moreover, user don't know anything about how
call are completed, nor how credentials are setup and soon.
In Kamailio, I guess that I have to use nat, tm and rtpproxy modules and
maybe uac. I am not sure how to setup it.
Can someone send me a clue?
Thank you,
Valter
14 Nov
14 Nov
12:04 a.m.
Am Freitag, 9. November 2018, 21:25:15 CET schrieb Valter Nogueira:
Today, I use Asterisk as a SIP/RTP PROXY
I proxy from customers Asterisks to a VOIP provider, in a multi-homed
server.
Now, I want to move to Kamailio without any rupture in customer's
configuration.
As anyone can imagine I am kind of lost.
USER ACCOUNTS
In Asterisk, I create a dynamic host account named ACCOUNT1 and I receive
in *FROM HEADER sip:ACCOUNT1@customer_ip_address*
In Kamailio, I have to define the account's domain like *kamctl add
ACCOUNT1(a)mydomain.com <ACCOUNT1(a)mydomain.com> password. *Kamailio just
accepts a REGISTER/INVITE from *ACCOUNT1(a)mydomain.com
<ACCOUNT1(a)mydomain.com>*
SIP/RTP PROXY
In Asterisk, I just dialout to the VOIP PROVIDER like *dial
(SIP/VOIP_ACCOUNT/${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
5:06 a.m.
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
12:43 p.m.
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>:
Am Freitag, 9. November 2018, 21:25:15 CET
schrieb Valter Nogueira:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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
2:39 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>:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Am Freitag, 9. November 2018, 21:25:15 CET
schrieb Valter Nogueira:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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
17 Nov
17 Nov
3:02 a.m.
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>
:
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>:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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
20 Nov
20 Nov
7:42 p.m.
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
<mailto:david.villasmil.work@gmail.com>
phone: +34669448337
ᐧ
On Wed, Nov 14, 2018 at 11:43 AM Valter Nogueira
<valter(a)fastway.com.br <mailto:valter@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
<mailto:david.villasmil.work@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(a)fastway.com.br <mailto:valter@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 <mailto:hw@kamailio.org>>:
Am Freitag, 9. November 2018, 21:25:15 CET schrieb
Valter Nogueira:
--
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
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
<mailto:ACCOUNT1@mydomain.com>
<ACCOUNT1(a)mydomain.com
<mailto:ACCOUNT1@mydomain.com>>
password. *Kamailio just
accepts a REGISTER/INVITE from
*ACCOUNT1(a)mydomain.com <mailto:ACCOUNT1@mydomain.com>
<ACCOUNT1(a)mydomain.com
<mailto:ACCOUNT1@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
<mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org <mailto:sr-users@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
21 Nov
21 Nov
3:40 a.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
2197
days inactive
2209
days old
7 comments
4 participants
participants (4)
-
Daniel-Constantin Mierla -
David Villasmil -
Henning Westerholt -
Valter Nogueira