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

[Matt Jordan (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24499?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Matt Jordan updated ASTERISK-24499:
-----------------------------------

    Target Release Version/s: 12.8.0

> 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: SVN, 12.6.1, 13.0.0
>            Reporter: Rusty Newton
>            Severity: Trivial
>      Target Release: 12.8.0
>
>
> 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 '912561234567'
> [Nov  5 09:31:38] DEBUG[3991][C-00000000]: pbx.c:4921 pbx_extension_helper: Launching 'Dial'
>     -- Executing [912561234567 at from-internal:1] Dial("PJSIP/6001-00000000", "PJSIP/mytrunk_endpoint/12561234567") 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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