I have a SIP TRUNK with TLS configured as PJSIP, the incoming calls are working but not the outgoing. I’m having this message.
Executing [s@macro-dialout-trunk:27] Dial(“PJSIP/1015-00000015”, “PJSIP/98092201111@+18093303000,300,T”) in new stack
[2025-11-17 14:58:24] ERROR[46756]: res_pjsip.c:852 ast_sip_create_dialog_uac: Endpoint ‘+18093303000’: Could not create dialog to invalid URI ‘+18093303000’. Is endpoint registered and reachable?
[2025-11-17 14:58:24] ERROR[46756]: chan_pjsip.c:2698 request: Failed to create outgoing session to endpoint ‘+18093303000’
[2025-11-17 14:58:24] NOTICE[51393][C-00000024]: app_dial.c:2709 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 [s@macro-dialout-trunk:28] NoOp(“PJSIP/1015-00000015”, “Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 3”) in new stack
This is the configuration in the file pjsip_additional.conf
thank you i’m very little experience with this but i’m trying to response those questions based on the behavior and config, please be patient with me
Does the TLS transport load and operate? The incoming calls are working, i’m assumig that the TLS are working, but do you have a command in order do i can verify this ?
Can the host be reached on the expected hostname and port using TLS? Yes the hostname can be reached.[root@issabel ~]# telnet isp.sip.com 5061Trying 25.25.25.25…Connected to isp.sip.com.Escape character is ‘^]’.
Does a packet capture show a connection attempt from Asterisk that fails? I do a debug and get only this message,
issabel*CLI> pjsip set logger onPJSIP Logging enabled== Using SIP RTP TOS bits 184== Using SIP RTP CoS mark 5> 0x7fce5c21a0e0 – Strict RTP learning after remote address set to: 192.168.2.18:4004– Executing [8092201111@from-internal:1] Dial(“SIP/1012-00000015”, “PJSIP/8092201111@isp.sip.com”) in new stack[2025-11-18 08:55:55] ERROR[50977]: chan_pjsip.c:2687 request: Unable to create PJSIP channel - endpoint ‘isp.sip.com’ was not found[2025-11-18 08:55:55] NOTICE[82222][C-00000037]: app_dial.c:2709 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 [8092201111@from-internal:2] Hangup(“SIP/1012-00000015”, “”) in new stack== Spawn extension (from-internal, 8092201111, 2) exited non-zero on ‘SIP/1012-00000015’– Executing [h@from-internal:1] Hangup(“SIP/1012-00000015”, “”) in new stack== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/1012-00000015’
TLS adds extra complexity as there are more moving parts, so with little experience it can be more difficult.
The “pjsip show outbound registrations” CLI command would show the outbound registration, and general console output would at least show that the correct endpoint for the incoming call is being used.
This isn’t like your previous log. If you don’t have an endpoint named “isp.sip.com” (which according to the log you don’t) then it won’t work.