PJSIP Trunking - Everyone is busy

Asterisk Asterisk SIP
1

I have Asterisk 18 installed on ubuntu 22.04. I configured my pjsip.conf and extensions.conf files using the instructions included here.

I get notified by asterisk CLI when 6001 registers and is “available” so clearly the softphone (Zoiper) is talking to the asterisk server, but, when I attempt an outbound call, I get an error which doesn’t seem so helpful that states “Everyone is busy”.

Can any experts look over the settings provided in the link above and make an educated guess what might be happening? Or perhaps, point me in the right direction to troubleshoot it better?

SIP debug output:

    -- Executing [1234567890@Zentrunk:1] Set("PJSIP/6001-0000000c", "CALLERID(all)="Plivo Number"<Plivo Number>") in new stack
    -- Executing [1234567890@Zentrunk:2] Dial("PJSIP/6001-0000000c", "PJSIP/1234567890@zentrunk_endpoint_out") in new stack
    -- Called PJSIP/1234567890@zentrunk_endpoint_out
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [1234567890@Zentrunk:3] Hangup("PJSIP/6001-0000000c", "") in new stack
  == Spawn extension (Zentrunk, 1234567890, 3) exited non-zero on 'PJSIP/6001-0000000c'
2

This is one of the reason why, the pjsip set logger on exist. Try it, and you will get the SIP final response of your carrier.

3

Thanks ambiorixg12. I already did that. The error posted above is the output from the PJSIP logger.

4

No is not, that is only regular CLI output

5

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