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 18:32:27 | Date Closed: | 2014-04-08 09:50:15 |
Priority: | Major | Regression? | Yes |
Status: | Closed/Complete | Components: | 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: |