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Summary:ASTERISK-23584: PJSIP 'Unable to create channel' when attempting to call from endpoint with UDP transport to one using WebSockets
Reporter:Rusty Newton (rnewton)Labels:
Date Opened:2014-04-03 18:32:27Date Closed:2014-04-08 09:50:15
Priority:MajorRegression?Yes
Status:Closed/CompleteComponents:Channels/chan_pjsip Resources/res_pjsip
Versions:SVN 12.1.1 Frequency of
Occurrence
Constant
Related
Issues:
Environment:Asterisk SVN-branch-12-r411670M WebRTC style configuration on LAN only. No STUN involved.Attachments:( 0) configs.txt
( 1) pjsip_debug.patch
Description: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@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
Comments: