No dial tone, only one way listening

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

There is nothing in that log which would generate dial tone. Did you mean ring back tone?

For SIP, dialtone is generally faked by the phone before there has been any interaction with the server.

I mean the ringtone that you hear as a caller on your phone when you call me. That has disappeared some time ago. Plus the other mistakes that the caller hears nothing - not even the ringtone. I can call out as normal.

The most likely reasons are either that the B side has stopped sending a ringing indication or has started sending it as early media. I’d start by explicitly calling Ringing before Dial, then try r on Dial, and also try Progress() before dial, with or without the preceding.