search for: devstate_event

Hi I have noticed that asterisk returns 'SIP 603' when the called party does not answer. My test setup is simple: two SIP phones (extensions: 100 and 111) registered to an Asterisk 1.8.30.0 gateway.The Dial timeout is 30 seconds. When 100 calls 111 and after 30 seconds, asterisk sends a CANCEL request to 111 (expected) and a '603 Decline' response to 100 (unexpected &

...o have stopped working when I upgraded to Asterisk 11.7. I get the following debug output, with a "no call pickup possible" message as soon as I press the '8': [Feb 12 18:17:39] -- Starting simple switch on 'DAHDI/2-1' [Feb 12 18:17:39] DEBUG[2339]: devicestate.c:442 devstate_event: device 'DAHDI/2' state '2' [Feb 12 18:17:42] DEBUG[5898][C-0000000e]: sig_analog.c:1600 analog_handle_dtmf: Begin DTMF digit: 0x2A '*' on DAHDI/2-1 [Feb 12 18:17:42] DEBUG[5898][C-0000000e]: chan_dahdi.c:2145 my_handle_dtmf: Begin DTMF digit: 0x2A '*' on DAHDI/2-1 [...

...set to 0,0 [Sep 14 08:21:11] DEBUG[9315]: devicestate.c:340 _ast_device_state: No provider found, checking channel drivers for DAHDI - 1 [Sep 14 08:21:11] DEBUG[9315]: devicestate.c:458 do_state_change: Changing state for DAHDI/1 - state 2 (In use) [Sep 14 08:21:11] DEBUG[9315]: devicestate.c:438 devstate_event: device 'DAHDI/1' state '2' [Sep 14 08:21:11] DEBUG[11186]: sig_analog.c:1769 __analog_ss_thread: __analog_ss_thread 1 -- Starting simple switch on 'DAHDI/1-1' [Sep 14 08:21:11] DEBUG[11186]: sig_analog.c:2392 __analog_ss_thread: Receiving DTMF cid on channel DAHDI/1-...