SRV Lookup for PJSIP trunk

So I’ve got a question about this. I built a SRV record and the system was able to identify the IPs (in order), as expected, but it can’t pass a call (receives a call though) - for outbound, it always considers the trunks unavailable. I make it an A RECORD, and it starts passing outbound calls - remove it so the SRV RECORD is all that’s left, and outbound fails… I was thinking, maybe in order to make it truly use SRV, you’d put something else in for the PORT (since the port is normally derived from the SRV record). Has anyone gotten this working in FPBX 14 with Asterisk 14, and if so, what needed to be done to make it work?


 Endpoint:  IO--VI_PJSIP                                         Unavailable   0 of inf
        Aor:  IO--VI_PJSIP                                       0
      Contact:  IO--VI_PJSIP/sip:trk-c1-alpha-diod-gen.gog ebb9bfa8a6 Unavail       0.000
  Transport:               udp      0      0
   Identify:  IO--VI_PJSIP/IO--VI_PJSIP

– Executing [[email protected]:23] GotoIf(“PJSIP/1602-00000023”, “0?customtrunk”) in new stack
– Executing [[email protected]:24] Dial(“PJSIP/1602-00000023”, “PJSIP/[email protected]–VI_PJSIP,300,T”) in
new stack
[2017-11-20 17:43:52] ERROR[20076]: res_pjsip.c:3106 ast_sip_create_dialog_uac: Endpoint ‘IO–VI_PJSIP’: Could not
create dialog to invalid URI ‘IO–VI_PJSIP’. Is endpoint registered and reachable?
[2017-11-20 17:43:52] ERROR[20076]: chan_pjsip.c:2178 request: Failed to create outgoing session to endpoint ‘IO–V
[2017-11-20 17:43:52] WARNING[31614][C-0000001d]: app_dial.c:2525 dial_exec_full: Unable to create channel of type
’PJSIP’ (cause 3 - No route to destination)

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