Send the invite to all asterisk, whichever has it will respond. On Thu, 25 Mar 2021 at 01:41, Vinicius Kwiecien Ruoso <[email protected]> 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 <[email protected]> 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 < > [email protected]> 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 > >> <[email protected]> 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:[email protected] > ;transport=ws > >> > > >> > None with success. Any hints on that can be wrong? I can share more > >> > detailed information. > >> > > >> > Greetings, > >> > Vinicius > >> _______________________________________________ > >> Kamailio (SER) - Users Mailing List > >> [email protected] > >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > _______________________________________________ > > Kamailio (SER) - Users Mailing List > > [email protected] > > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > _______________________________________________ > Kamailio (SER) - Users Mailing List > [email protected] > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > -- Regards, David Villasmil email: [email protected] phone: +34669448337
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