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@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@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@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-path/74079
>> >
>> > 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@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@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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>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.work@gmail.com
> phone: +34669448337
> _______________________________________________
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