Yes, *add_contact_alias* for register on the way to asterisk followed by
*handle_ruri_alias* for requests from asterisk will do the job.
I assume you followed the famous
Regards,
--Sergiu
On Fri, Mar 26, 2021 at 11:04 AM Vinicius Kwiecien Ruoso <
vinicius(a)leads2b.com> wrote:
Asterisk 13 will be EOL soon, so that would be my last
resort.
Do you have any input if the contact alias should be enough to get
this working in that case?
On Thu, Mar 25, 2021 at 9:47 AM Sergiu Pojoga <pojogas(a)gmail.com> wrote:
Use asterisk 13 with chan_sip + PATH
Regardless, it's becoming off-topic as the quarrel is with Asterisk.
On Thu, Mar 25, 2021 at 8:16 AM Vinicius Kwiecien Ruoso <
vinicius(a)leads2b.com> wrote:
>
> Hi David,
>
> Each Asterisk is independent in my setup. Also, the issue is with
> calls from Asterisk with destination to a client connected via
> websocket. The browser is not directly connected to Asterisk, but to
> Kamailio.
>
> Thanks,
> Vinicius
>
> On Thu, Mar 25, 2021 at 7:01 AM David Villasmil
> <david.villasmil.work(a)gmail.com> wrote:
> >
> > Send the invite to all asterisk, whichever has it will respond.
> >
> > On Thu, 25 Mar 2021 at 01:41, Vinicius Kwiecien Ruoso <
vinicius(a)leads2b.com> wrote:
> >>
> >> Hi Segiu,
> >>
> >> Yeah, I understand that won't be the way forward, so I need another
> >> way for Kamailio to figure out the route to the websocket when
> >> Asterisk is making the call.
> >>
> >> I've shared my config file, also with some attempts commented in the
> >> previous emails. The alternatives were to use an alias, or an encoded
> >> param. I think the alias would be a better alternative, but could not
> >> get it right yet.
> >>
> >> Greetings,
> >> Vinicius
> >>
> >> On Wed, Mar 24, 2021 at 7:16 PM Sergiu Pojoga <pojogas(a)gmail.com>
wrote:
> >> >
> >> > Vinicius,
> >> >
> >> > The obvious is that PATH is broken in Asterisk's PJSIP and they
won't do anything about it as it's marked "minor". It's been
2 yrs now that
me and others have reported it.
> >> > [3]
https://community.asterisk.org/t/wrong-d-uri-for-invites-with-pjsip-and-pat…
> >> >
> >> > Bottom line: there's nothing wrong with Kamailio. Start looking
for a workaround, don't bet on Sangoma to fix it any time soon, lol.
> >> >
> >> > Regards,
> >> > --Sergiu
> >> >
> >> >
> >> >
> >> > On Wed, Mar 24, 2021 at 4:52 PM Vinicius Kwiecien Ruoso <
vinicius(a)leads2b.com> wrote:
> >> >>
> >> >> Hi!
> >> >>
> >> >> Thanks for the fast response. Sorry about not replying to the
correct
> >> >> email, I've just
entered the list and was not getting its
individual
> >> >> emails.
> >> >>
> >> >> > So it’s an outbound call to a webrtc registered user? If so,
kamailio
> >> >> > should route it to
wherever the called user is registered.
> >> >>
> >> >> Yes, it is a call from the backend Asterisk to a user registered
via
> >> >> the websocket. The register
is not stored in Kamailio, so it
needs to
> >> >> use the information in the
INVITE message to be able to route to
the
> >> >> correct connection.
> >> >>
> >> >> > you'll need to share your config and logs. This should
work in
your scenario.
> >> >>
> >> >> Turns out looking closer, Asterisk is not respecting the Path
protocol
> >> >> [1] [2]. In the INVITE sent
to Kamailio, there is no information
about
> >> >> the path in the message.
> >> >>
> >> >> To me the best second approach that should work is the
"alias="
> >> >> information in the contact. That makes sense?
> >> >>
> >> >> I'm sharing my current config as an attachment here. I'm
new to
> >> >> Kamailio, so I might be missing something really obvious here.
> >> >>
> >> >> [1]
https://issues.asterisk.org/jira/browse/ASTERISK-28211
> >> >> [2]
https://community.asterisk.org/t/pjsip-path-module-issues/88046/12
> >> >>
> >> >> Thanks,
> >> >> Vinicius
> >> >>
> >> >> On Tue, Mar 23, 2021 at 6:43 PM Vinicius Kwiecien Ruoso
> >> >> <vinicius(a)leads2b.com> wrote:
> >> >> >
> >> >> > Hi!
> >> >> >
> >> >> > I'm using Kamaio in front of multiple Asterisk instances.
At
this
> >> >> > moment it works as a
SIP over Websocket proxy, with rtpengine,
for
> >> >> > browser clients to
connect to Asterisk using WebRTC. I do not
use the
> >> >> > registration module of
Kamailio, as each backend Asterisk is
> >> >> > independent and handles its own registrations.
> >> >> >
> >> >> > Everything works great when making calls from the browser, and
the
> >> >> > routing is correctly
executed by Kamailio based on the request
SIP
> >> >> > domain. We have an
internal routing API that it calls to
discover
> >> >> > which backend Asterisk
to route the calls.
> >> >> >
> >> >> > The issue I have is when a call initiates from that backend
Asterisk,
> >> >> > trying to reach a
contact that is connected in Kamailio via the
> >> >> > websocket. The Asterisk sends the message to the proxy, and
Kamailio
> >> >> > must route it to the
corresponding websocket.
> >> >> >
> >> >> > I've tried a few approaches:
> >> >> > - using add_contact_alias + handle_ruri_alias: I have the
alias
with
> >> >> >
alias=<ip>~<port>~ws in the contact registration, but for some
reason
> >> >> > handle_ruri_alias
cannot use it
> >> >> > - using the Path module on Asterisk, so when registering, the
path is
> >> >> > recorded and sent back
from Asterisk, Kamailio is also not
respecting
> >> >> > that
> >> >> > - Using contact_param_encode and contact_param_encode and
> >> >> > contact_param_decode_ruri, but the encoded sip address is
always the
> >> >> > invalid websocket,
like sip:58c0ktrg@5hp0nn5hqqv9.invalid
;transport=ws
> >> >> >
> >> >> > None with success. Any hints on that can be wrong? I can share
more
> >> > detailed information.
> >> >
> >> > Greetings,
> >> > Vinicius
> >> _______________________________________________
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> >
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--
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
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