25 Mar
2021
25 Mar
'21
3:44 p.m.
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:
--
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
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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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