Citation Envoyé par jean Voir le message
il faudrait les traces (avec core set verbose 3 mini) de la console, et idéalement, les premières trames émises au début d'un appel avec rtp set debug on
Voici les traces:

Code:
=========================================================================
Connected to Asterisk 13.7.0 currently running on srvasterisk01 (pid = 3272)
  == Using SIP RTP CoS mark 5
    -- Executing [100@global:1] Dial("SIP/101-00000000", "SIP/100") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/100
    -- SIP/100-00000001 is ringing
    -- Registered SIP '101' at 88.xxxxxxx:36996
       > Saved useragent "Z 3.9.32144 r32121" for peer 101
[Feb  8 22:39:53] NOTICE[3779]: chan_sip.c:27701 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 101
[Feb  8 22:39:56] NOTICE[3779]: chan_sip.c:29374 sip_poke_noanswer: Peer '101' is now UNREACHABLE!  Last qualify: 0
    -- SIP/100-00000001 answered SIP/101-00000000
    -- Channel SIP/100-00000001 joined 'simple_bridge' basic-bridge <801e5ab1-7e62-4589-bef5-3f58cf10efba>
    -- Channel SIP/101-00000000 joined 'simple_bridge' basic-bridge <801e5ab1-7e62-4589-bef5-3f58cf10efba>
       > 0x7f42b400bca0 -- Probation passed - setting RTP source address to 192.xxxxxx:8000
       > 0x2002a00 -- Probation passed - setting RTP source address to 192.xxxxxx:8000
[Feb  8 22:40:06] NOTICE[3779]: chan_sip.c:23938 handle_response_peerpoke: Peer '101' is now Reachable. (5ms / 2000ms)
[Feb  8 22:40:18] NOTICE[3779]: chan_sip.c:27701 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 101
[Feb  8 22:40:34] WARNING[3779]: chan_sip.c:4015 retrans_pkt: Retransmission timeout reached on transmission Y1gGaU5kx2gI9a_OTXTlSw.. for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32001ms with no response
[Feb  8 22:40:34] WARNING[3779]: chan_sip.c:4044 retrans_pkt: Hanging up call Y1gGaU5kx2gI9a_OTXTlSw.. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
    -- Channel SIP/101-00000000 left 'simple_bridge' basic-bridge <801e5ab1-7e62-4589-bef5-3f58cf10efba>
  == Spawn extension (global, 100, 1) exited non-zero on 'SIP/101-00000000'
    -- Channel SIP/100-00000001 left 'simple_bridge' basic-bridge <801e5ab1-7e62-4589-bef5-3f58cf10efba>
srvasterisk01*CLI>