Nick Awesome wrote:
Hi all, In my case I using realtime, here is how it looks in plant [10001] type=registration transport=upd_static outbound_auth=10001 server_uri=sip:600@192.168.1.1:5060 client_uri=sip:600@192.168.1.4:5060 [10001] type=auth auth_type=userpass password=600 username=600 [10001] type=aor contact=sip:192.168.1.4:5060 [10001] type=endpoint transport=upd_static context=dialmap disallow=all allow=ulaw outbound_auth=10001 aors=10001 [10001] type=identify endpoint=10001 match=192.168.1.1 when I call 600 from other pbx I getting an notice NOTICE[10202]: res_pjsip/pjsip_distributor.c:246 log_unidentified_request: Request from '"Ilya"<sip:502@192.168.1.1>' failed for '192.168.1.1:5060' (callid: ZTNhYjU4ZjU5ZmUxNjM5M2FlYjBlYTE3YzgwZTU4MGY.) - No matching endpoint found and "Not Accessable" on phone let's imagine that 600 its external number of voip operator, and I wanna accept all incoming calls from it (no matter what caller id it has) what I doing wrong?
When receiving calls from a VoIP provider you have to match using the source IP address. You also don't authenticate as the provider will refuse to do so.
When you control both ends it's really up to you whether to do the matching based on the source IP address OR use a user account with authentication. If using the user account the user portion of the From header has to be set to the username (from_user in pjsip, fromuser in chan_sip).
Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users