[asterisk-bugs] [JIRA] (ASTERISK-24499) Need more explicit debug when PJSIP dialstring is invalid

[Rusty Newton (JIRA)]

Rusty Newton created ASTERISK-24499:
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             Summary: Need more explicit debug when PJSIP dialstring is invalid
                 Key: ASTERISK-24499
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24499
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Applications/app_dial, Resources/res_pjsip
    Affects Versions: 13.0.0, 12.6.1, SVN
            Reporter: Rusty Newton
            Severity: Trivial


I forgot how to form a PJSIP dialstring and wrote the one below:
{noformat}
exten => _91NXXXXXXXXX,1,Dial(PJSIP/mytrunk_endpoint/${EXTEN:1})
{noformat}
Where as I should have written:
{noformat}
exten => _91NXXXXXXXXX,1,Dial(PJSIP/${EXTEN:1}@mytrunk_endpoint)
{noformat}

The issue is console logger output didn't help me figure out what I did wrong.

With all log channels and verbosity up you see the following:

{noformat}
[Nov  5 09:31:38] DEBUG[3871]: chan_pjsip.c:2118 pbx_start_incoming_request: Started PBX on new PJSIP channel PJSIP/6001-00000000
[Nov  5 09:31:38] DEBUG[3991][C-00000000]: pbx.c:3723 ast_str_retrieve_variable: Result of 'EXTEN' is '912564799317'
[Nov  5 09:31:38] DEBUG[3991][C-00000000]: pbx.c:4921 pbx_extension_helper: Launching 'Dial'
    -- Executing [912564799317 at from-internal:1] Dial("PJSIP/6001-00000000", "PJSIP/mytrunk_endpoint/12564799317") in new stack
[Nov  5 09:31:38] WARNING[3991][C-00000000]: app_dial.c:2431 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)
[Nov  5 09:31:38] DEBUG[3991][C-00000000]: app_dial.c:3090 dial_exec_full: Exiting with DIALSTATUS=CHANUNAVAIL.
    -- Auto fallthrough, channel 'PJSIP/6001-00000000' status is 'CHANUNAVAIL'
[Nov  5 09:31:38] DEBUG[3871]: res_pjsip_session.c:1852 handle_outgoing_response: Method is INVITE, Response is 503 Service Unavailable
{noformat}

I'm told Asterisk can't create a channel of type PJSIP and I can see that a 503 goes out to the originating channel.

Fortunately I realized after this what I did wrong as I've done this a few times before, however it could be frustrating for those moving from chan_sip to res_pjsip. I'm wondering if there is a way we could get a more explicit message to help out the user.



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