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 >
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