[asterisk-bugs] [JIRA] (ASTERISK-23584) PJSIP 'Unable to create channel' when attempting to call from endpoint with UDP transport to one using WebSockets

[Rusty Newton (JIRA)]

Rusty Newton created ASTERISK-23584:
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             Summary: PJSIP 'Unable to create channel' when attempting to call from endpoint with UDP transport to one using WebSockets
                 Key: ASTERISK-23584
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23584
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Channels/chan_pjsip, Resources/res_pjsip
    Affects Versions: 12.1.1, SVN
         Environment: Asterisk SVN-branch-12-r411670M
            Reporter: Rusty Newton


Found while testing WebRTC style configuration with a SIPML5 endpoint
Configure two endpoints:

Endpoint A: Uses a UDP transport
Endpoint B: Uses a WebSockets transport (and other configuration as shown in config.txt)

Call from A to B, cal fails with:
{noformat}
[Apr  3 18:30:44] WARNING[30845][C-00000002]: app_dial.c:2428 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)
{noformat}

Mark Michelson provided a debug patch so we can see a little more what is happening:
{noformat}
    -- Executing [6002 at from-internal:1] Dial("PJSIP/6001-00000002", "PJSIP/6002,15") in new stack
[Apr  3 18:30:44] NOTICE[30469]: res_pjsip_session.c:1209 ast_sip_session_create_outgoing: Using AOR list from endpoint as basis for location
[Apr  3 18:30:44] NOTICE[30469]: res_pjsip.c:1527 ast_sip_create_dialog_uac: Failed to get transport selector for this endpoint
[Apr  3 18:30:44] NOTICE[30469]: res_pjsip_session.c:1237 ast_sip_session_create_outgoing: Error creating SIP dialog
[Apr  3 18:30:44] WARNING[30845][C-00000002]: app_dial.c:2428 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)
{noformat}

I will attach the pjsip.conf config in config.txt and Mark's patch in pjsip_debug.patch



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