Hello together,
I have an Asterisk 14.2.0 installation, in which suddenly my callers no longer hear a dial tone, they no longer hear the voicebox announcement, even though my phone rings.
I hear the caller speaking, but the caller cannot hear me. Enclosed the log from the Asterisk CLI of such a call.
Connected to Asterisk 14.2.0-rc2 currently running on dianibeach (pid = 591)
-- Executing [070321229517@telekom_in:1] GotoIf("PJSIP/telekom_in-00000006", "1?unavail89517") in new stack
-- Goto (telekom_in,070321229517,3)
-- Executing [070321229517@telekom_in:3] Playback("PJSIP/telekom_in-00000006", "privat-Tag") in new stack
-- <PJSIP/telekom_in-00000006> Playing 'privat-Tag.slin' (language 'de')
> 0x2aeb940 -- Probation passed - setting RTP source address to 217.0.5.151:31354
-- Executing [070329063112@telekom_in:1] GotoIf("PJSIP/telekom_in-00000007", "0?unavail63112") in new stack
-- Executing [070329063112@telekom_in:2] GotoIfTime("PJSIP/telekom_in-00000007", "16:00-17:29,fri,*,*?friday-closed") in new stack
-- Executing [070329063112@telekom_in:3] GotoIfTime("PJSIP/telekom_in-00000007", "18:00-7:59,mon-fri,*,*?unavail63112") in new stack
[2020-07-24 12:33:11] WARNING[3643][C-00000005]: chan_sip.c:22876 func_header_read: This function can only be used on SIP channels.
-- Executing [070329063112@telekom_in:4] Verbose("PJSIP/telekom_in-00000007", "Header To: ,1") in new stack
[2020-07-24 12:33:11] WARNING[3643][C-00000005]: app_verbose.c:101 verbose_exec: 'Header To: ' is not a verboser number
1
-- Executing [070329063112@telekom_in:5] Dial("PJSIP/telekom_in-00000007", "PJSIP/63112&PJSIP/6001,20,m[To_of_us]") in new stack
[2020-07-24 12:33:11] ERROR[3636]: res_pjsip.c:2887 ast_sip_create_dialog_uac: Could not create dialog to endpoint '6001' as URI '6001' is not valid
[2020-07-24 12:33:11] ERROR[3636]: chan_pjsip.c:2146 request: Failed to create outgoing session to endpoint '6001'
[2020-07-24 12:33:11] WARNING[3643][C-00000005]: app_dial.c:2530 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
-- Called PJSIP/63112
-- Started music on hold, class 'default', on channel 'PJSIP/telekom_in-00000007'
-- PJSIP/63112-00000008 is ringing
-- PJSIP/63112-00000008 answered PJSIP/telekom_in-00000007
-- Stopped music on hold on PJSIP/telekom_in-00000007
-- Channel PJSIP/63112-00000008 joined 'simple_bridge' basic-bridge <edeff7ca-9d0e-4372-89a1-a401fb8e8053>
-- Channel PJSIP/telekom_in-00000007 joined 'simple_bridge' basic-bridge <edeff7ca-9d0e-4372-89a1-a401fb8e8053>
> 0x2ad5238 -- Probation passed - setting RTP source address to 192.168.0.24:5010
> 0x2ae1980 -- Probation passed - setting RTP source address to 217.0.5.180:64576
-- Channel PJSIP/telekom_in-00000007 left 'simple_bridge' basic-bridge <edeff7ca-9d0e-4372-89a1-a401fb8e8053>
-- Channel PJSIP/63112-00000008 left 'simple_bridge' basic-bridge <edeff7ca-9d0e-4372-89a1-a401fb8e8053>
== Spawn extension (telekom_in, 070329063112, 5) exited non-zero on 'PJSIP/telekom_in-00000007'
What can that be? The system had worked completely before.
Best regards
Andreas