I’ve followed the asterisk rebuild process presented for IncrediblePBX from page 26267 of the nerdvittles site (can’t post the link).
Everything compiles without error, but I don’t have a working system. If anybody can explain why I can’t place an outgoing call, I’d appreciated it:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
> 0xb6324220 – Strict RTP learning after remote address set to: 192.168.2.21:5004
– Executing [6085------@from-internal:1] Set(“SIP/1024-00000003”, “CALLERID(dnid)=6085------”) in new stack
– Executing [6085------@from-internal:2] Set(“SIP/1024-00000003”, “CALLERID(dnid)=16085------”) in new stack
– Executing [6085------7@from-internal:3] Goto(“SIP/1024-00000003”, “16085------,1”) in new stack
– Goto (from-internal,16085------,1)
– Executing [16085------@from-internal:1] Set(“SIP/1024-00000003”, “CHANNEL(accountcode)=Google Voice”) in new stack
– Executing [1608------@from-internal:2] Dial(“SIP/1024-00000003”, “PJSIP/16085------@gvsip,r”) in new stack
[2018-07-04 18:53:17] ERROR[3286]: chan_pjsip.c:2215 request: Unable to create PJSIP channel - endpoint ‘gvsip’ was not found
[2018-07-04 18:53:17] WARNING[3285][C-00000002]: app_dial.c:2527 dial_exec_full: Unable to create channel of type ‘PJSIP’ (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [16085------@from-internal:3] Progress(“SIP/1024-00000003”, “”) in new stack
– Executing [16085------@from-internal:4] Wait(“SIP/1024-00000003”, “1”) in new stack
> 0xb6324220 – Strict RTP switching to RTP target address 192.168.2.21:5004 as source
– Executing [16085------@from-internal:5] Playback(“SIP/1024-00000003”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
– Playing ‘silence/1.ulaw’ (language ‘en’)
– Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘en’)
– Playing ‘check-number-dial-again.ulaw’ (language ‘en’)