Probleme mit Asterisk und Toplink

Toggle sidebar Toggle sidebar
Upgrade 3CX to v18 and get it hosted free!
L

Lapidu

Neuer User
Mitglied seit
29 Mrz 2006
Beiträge
48
Punkte für Reaktionen
0
Punkte
6
Hallo zusammen,

ich betreibe einen Asterisk in Verbindung mit Toplink Trunk.

Seit ca 3 Wochen habe ich das Problem das wenn eine Nummer angerufen wird und die Nebenstelle dran geht die Verbindung sofort abbricht. Ich habe auch direkt eine Toplinknummer die direkt auf Ansage auf meinem Asterisk geht. Das funktioniert.

Anbei habe ich mal den Log nach dem klingeln der Nebenstelle kopiert.

Code: 
-- Called SIP/218
    -- SIP/218-00000006 is ringing
    -- SIP/218-00000006 answered SIP/Toplink-00000005
    -- Channel SIP/218-00000006 joined 'simple_bridge' basic-bridge <2afdc732-ece8-43f1-8fb4-f907af33d601>
    -- Channel SIP/Toplink-00000005 joined 'simple_bridge' basic-bridge <2afdc732-ece8-43f1-8fb4-f907af33d601>
       > 0x7f669c00cf00 -- Probation passed - setting RTP source address to 91.16.229.35:7086
       > 0x7f66b0022b50 -- Probation passed - setting RTP source address to 213.218.12.40:11630
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
    -- Started music on hold, class 'default', on channel 'SIP/218-00000006'
       > 0x7f66b0022b50 -- Probation passed - setting RTP source address to 213.218.12.40:11630
       > 0x7f66b0022b50 -- Probation passed - setting RTP source address to 213.218.12.40:11630
    -- Stopped music on hold on SIP/218-00000006
       > 0x7f66b0022b50 -- Probation passed - setting RTP source address to 213.218.12.40:11630
       > 0x7f669c00cf00 -- Probation passed - setting RTP source address to 91.16.229.35:7086
    -- Channel SIP/Toplink-00000005 left 'simple_bridge' basic-bridge <2afdc732-ece8-43f1-8fb4-f907af33d601>
  == Spawn extension (handle_inbound, 1, 43) exited non-zero on 'SIP/Toplink-00000005'
    -- Channel SIP/218-00000006 left 'simple_bridge' basic-bridge <2afdc732-ece8-43f1-8fb4-f907af33d601>
    -- Executing [h@handle_inbound:1] System("SIP/Toplink-00000005", "/etc/asterisk/consolidatecdr.sh v22017024287244466-1510651461.5 "ANSWER" &") in new stack
    -- Executing [h@handle_inbound:2] GotoIf("SIP/Toplink-00000005", "1?3:5") in new stack
    -- Goto (handle_inbound,h,3)
  == MixMonitor close filestream (mixed)
    -- Executing [h@handle_inbound:4] Set("SIP/Toplink-00000005", "__HASMONITOR=") in new stack
    -- Executing [h@handle_inbound:5] NoOp("SIP/Toplink-00000005", "Finish if_handle_inbound_71") in new stack
    -- Executing [h@handle_inbound:6] Hangup("SIP/Toplink-00000005", "") in new stack
  == Spawn extension (handle_inbound, h, 6) exited non-zero on 'SIP/Toplink-00000005'

//edit stoney: solche Blöcke bitte in [CODE] setzen [/CODE]
//edit2 stoney bitte keine Doppelposts

### Zussmmenführung ###

Das sagt der SIP Debug

Code: 
    -- Executing [SIP-User@inc_sip:1] Set("SIP/Toplink-0000002b", "DEST=004989919291091") in new stack
    -- Executing [SIP-User@inc_sip:2] NoOp("SIP/Toplink-0000002b", "Eingehender Ruf von Provider Toplink Multichannel fuer 004989919291091") in new stack
    -- Executing [SIP-User@inc_sip:3] Set("SIP/Toplink-0000002b", "CDR(amaflags)=Documentation") in new stack
[2017-11-14 13:40:03] WARNING[1393]: func_cdr.c:404 cdr_write_callback: Using the CDR function to set 'amaflags' is deprecated. Please use the CHANNEL function instead.
    -- Executing [SIP-User@inc_sip:4] Set("SIP/Toplink-0000002b", "CDR(provider)=Toplink") in new stack
    -- Executing [SIP-User@inc_sip:5] Set("SIP/Toplink-0000002b", "line=intern") in new stack
    -- Executing [SIP-User@inc_sip:6] GotoIf("SIP/Toplink-0000002b", "1?7:8") in new stack
    -- Goto (inc_sip,SIP-User,7)
    -- Executing [SIP-User@inc_sip:7] Set("SIP/Toplink-0000002b", "CALLERID(num)=XXXXXXXXXX") in new stack
    -- Executing [SIP-User@inc_sip:8] NoOp("SIP/Toplink-0000002b", "Finish if_inc_sip_552") in new stack
    -- Executing [SIP-User@inc_sip:9] GotoIf("SIP/Toplink-0000002b", "0?10:12") in new stack
    -- Goto (inc_sip,SIP-User,12)
    -- Executing [SIP-User@inc_sip:12] GotoIf("SIP/Toplink-0000002b", "0?13:15") in new stack
    -- Goto (inc_sip,SIP-User,15)
    -- Executing [SIP-User@inc_sip:15] GotoIf("SIP/Toplink-0000002b", "0?16:17") in new stack
    -- Goto (inc_sip,SIP-User,17)
    -- Executing [SIP-User@inc_sip:17] NoOp("SIP/Toplink-0000002b", "Finish if_if_if_inc_sip_553_554_555") in new stack
    -- Executing [SIP-User@inc_sip:18] NoOp("SIP/Toplink-0000002b", "Finish if_if_inc_sip_553_554") in new stack
    -- Executing [SIP-User@inc_sip:19] NoOp("SIP/Toplink-0000002b", "Finish if_inc_sip_553") in new stack
    -- Executing [SIP-User@inc_sip:20] GotoIf("SIP/Toplink-0000002b", "1?21:25") in new stack
    -- Goto (inc_sip,SIP-User,21)
    -- Executing [SIP-User@inc_sip:21] Set("SIP/Toplink-0000002b", "CDR(CalledNumber)=XXXXXXXXXX") in new stack
    -- Executing [SIP-User@inc_sip:22] Set("SIP/Toplink-0000002b", "nst=Toplink_919291091") in new stack
    -- Executing [SIP-User@inc_sip:23] Goto("SIP/Toplink-0000002b", "XXXXXXXXXX,1") in new stack
    -- Goto (inc_sip,XXXXXXXXXX,1)
    -- Executing [XXXXXXXXXX@inc_sip:1] NoOp("SIP/Toplink-0000002b", "Eingehender Ruf von Provider Toplink Multichannel (Aldo) fuer XXXXXXXXXX") in new stack
    -- Executing [XXXXXXXXXX@inc_sip:2] Set("SIP/Toplink-0000002b", "CDR(amaflags)=Documentation") in new stack
[2017-11-14 13:40:03] WARNING[1393]: func_cdr.c:404 cdr_write_callback: Using the CDR function to set 'amaflags' is deprecated. Please use the CHANNEL function instead.
    -- Executing [XXXXXXXXXX@inc_sip:3] Set("SIP/Toplink-0000002b", "CDR(CalledNumber)=XXXXXXXXXX") in new stack
    -- Executing [XXXXXXXXXX@inc_sip:4] Set("SIP/Toplink-0000002b", "CDR(provider)=Toplink") in new stack
    -- Executing [XXXXXXXXXX@inc_sip:5] Set("SIP/Toplink-0000002b", "nst=Toplink_919291091") in new stack
    -- Executing [XXXXXXXXXX@inc_sip:6] Set("SIP/Toplink-0000002b", "line=intern") in new stack
    -- Executing [XXXXXXXXXX@inc_sip:7] Set("SIP/Toplink-0000002b", "nsti=300") in new stack
    -- Executing [XXXXXXXXXX@inc_sip:8] Set("SIP/Toplink-0000002b", "ToDial=SIP/300") in new stack
    -- Executing [XXXXXXXXXX@inc_sip:9] Set("SIP/Toplink-0000002b", "dialopt=twhr") in new stack
    -- Executing [XXXXXXXXXX@inc_sip:10] Goto("SIP/Toplink-0000002b", "handle_inbound,s,1") in new stack
    -- Goto (handle_inbound,s,1)
    -- Executing [s@handle_inbound:1] GotoIf("SIP/Toplink-0000002b", "1?2:18") in new stack
    -- Goto (handle_inbound,s,2)
    -- Executing [s@handle_inbound:2] GotoIf("SIP/Toplink-0000002b", "0?3:4") in new stack
    -- Goto (handle_inbound,s,4)
    -- Executing [s@handle_inbound:4] NoOp("SIP/Toplink-0000002b", "Finish if_if_handle_inbound_419_420") in new stack
    -- Executing [s@handle_inbound:5] GotoIf("SIP/Toplink-0000002b", "0?6:8") in new stack
    -- Goto (handle_inbound,s,8)
    -- Executing [s@handle_inbound:8] Set("SIP/Toplink-0000002b", "CDR(int_exten)=300") in new stack
    -- Executing [s@handle_inbound:9] Set("SIP/Toplink-0000002b", "CDR(ext_exten)=XXXXXXXXXX") in new stack
    -- Executing [s@handle_inbound:10] Set("SIP/Toplink-0000002b", "CDR(direction)=Ankommend") in new stack
    -- Executing [s@handle_inbound:11] NoOp("SIP/Toplink-0000002b", "Finish if_if_handle_inbound_419_421") in new stack
    -- Executing [s@handle_inbound:12] Gosub("SIP/Toplink-0000002b", "check_callerid,s,1") in new stack
    -- Executing [s@check_callerid:1] GotoIf("SIP/Toplink-0000002b", "0?2:4") in new stack
    -- Goto (check_callerid,s,4)
    -- Executing [s@check_callerid:4] Set("SIP/Toplink-0000002b", "LOCAL(Max)=15") in new stack
    -- Executing [s@check_callerid:5] MSet("SIP/Toplink-0000002b", "LOCAL(x)=14") in new stack
    -- Executing [s@check_callerid:6] GotoIf("SIP/Toplink-0000002b", "1?7:13") in new stack
    -- Goto (check_callerid,s,7)
    -- Executing [s@check_callerid:7] Set("SIP/Toplink-0000002b", "LOCAL(Test)=XXXXXXXXXX0") in new stack
    -- Executing [s@check_callerid:8] GotoIf("SIP/Toplink-0000002b", "0?9:10") in new stack
    -- Goto (check_callerid,s,10)
    -- Executing [s@check_callerid:10] NoOp("SIP/Toplink-0000002b", "Finish if_for_if_check_callerid_413_414_415") in new stack
    -- Executing [s@check_callerid:11] MSet("SIP/Toplink-0000002b", "LOCAL(x)=13") in new stack
    -- Executing [s@check_callerid:12] Goto("SIP/Toplink-0000002b", "6") in new stack
    -- Goto (check_callerid,s,6)
    -- Executing [s@check_callerid:6] GotoIf("SIP/Toplink-0000002b", "1?7:13") in new stack
    -- Goto (check_callerid,s,7)
    -- Executing [s@check_callerid:7] Set("SIP/Toplink-0000002b", "LOCAL(Test)=XXXXXXXXXX") in new stack
    -- Executing [s@check_callerid:8] GotoIf("SIP/Toplink-0000002b", "0?9:10") in new stack
    -- Goto (check_callerid,s,10)
    -- Executing [s@check_callerid:10] NoOp("SIP/Toplink-0000002b", "Finish if_for_if_check_callerid_413_414_415") in new stack
    -- Executing [s@check_callerid:11] MSet("SIP/Toplink-0000002b", "LOCAL(x)=12") in new stack
    -- Executing [s@check_callerid:12] Goto("SIP/Toplink-0000002b", "6") in new stack
    -- Goto (check_callerid,s,6)
    -- Executing [s@check_callerid:6] GotoIf("SIP/Toplink-0000002b", "1?7:13") in new stack
    -- Goto (check_callerid,s,7)
    -- Executing [s@check_callerid:7] Set("SIP/Toplink-0000002b", "LOCAL(Test)=XXXXXXXXXX3") in new stack
    -- Executing [s@check_callerid:8] GotoIf("SIP/Toplink-0000002b", "0?9:10") in new stack
    -- Goto (check_callerid,s,10)
    -- Executing [s@check_callerid:10] NoOp("SIP/Toplink-0000002b", "Finish if_for_if_check_callerid_413_414_415") in new stack
    -- Executing [s@check_callerid:11] MSet("SIP/Toplink-0000002b", "LOCAL(x)=11") in new stack
    -- Executing [s@check_callerid:12] Goto("SIP/Toplink-0000002b", "6") in new stack
    -- Goto (check_callerid,s,6)
    -- Executing [s@check_callerid:6] GotoIf("SIP/Toplink-0000002b", "1?7:13") in new stack
    -- Goto (check_callerid,s,7)
    -- Executing [s@check_callerid:7] Set("SIP/Toplink-0000002b", "LOCAL(Test)=XXXXXXXXXX") in new stack
    -- Executing [s@check_callerid:8] GotoIf("SIP/Toplink-0000002b", "0?9:10") in new stack
    -- Goto (check_callerid,s,10)
    -- Executing [s@check_callerid:10] NoOp("SIP/Toplink-0000002b", "Finish if_for_if_check_callerid_413_414_415") in new stack
    -- Executing [s@check_callerid:11] MSet("SIP/Toplink-0000002b", "LOCAL(x)=10") in new stack
    -- Executing [s@check_callerid:12] Goto("SIP/Toplink-0000002b", "6") in new stack
    -- Goto (check_callerid,s,6)
    -- Executing [s@check_callerid:6] GotoIf("SIP/Toplink-0000002b", "1?7:13") in new stack
    -- Goto (check_callerid,s,7)
    -- Executing [s@check_callerid:7] Set("SIP/Toplink-0000002b", "LOCAL(Test)=0049152539") in new stack
    -- Executing [s@check_callerid:8] GotoIf("SIP/Toplink-0000002b", "0?9:10") in new stack
    -- Goto (check_callerid,s,10)
    -- Executing [s@check_callerid:10] NoOp("SIP/Toplink-0000002b", "Finish if_for_if_check_callerid_413_414_415") in new stack
    -- Executing [s@check_callerid:11] MSet("SIP/Toplink-0000002b", "LOCAL(x)=9") in new stack
    -- Executing [s@check_callerid:12] Goto("SIP/Toplink-0000002b", "6") in new stack
    -- Goto (check_callerid,s,6)
    -- Executing [s@check_callerid:6] GotoIf("SIP/Toplink-0000002b", "1?7:13") in new stack
    -- Goto (check_callerid,s,7)
    -- Executing [s@check_callerid:7] Set("SIP/Toplink-0000002b", "LOCAL(Test)=004915253") in new stack
    -- Executing [s@check_callerid:8] GotoIf("SIP/Toplink-0000002b", "0?9:10") in new stack
    -- Goto (check_callerid,s,10)
    -- Executing [s@check_callerid:10] NoOp("SIP/Toplink-0000002b", "Finish if_for_if_check_callerid_413_414_415") in new stack
    -- Executing [s@check_callerid:11] MSet("SIP/Toplink-0000002b", "LOCAL(x)=8") in new stack
    -- Executing [s@check_callerid:12] Goto("SIP/Toplink-0000002b", "6") in new stack
    -- Goto (check_callerid,s,6)
    -- Executing [s@check_callerid:6] GotoIf("SIP/Toplink-0000002b", "1?7:13") in new stack
    -- Goto (check_callerid,s,7)
    -- Executing [s@check_callerid:7] Set("SIP/Toplink-0000002b", "LOCAL(Test)=00491525") in new stack
    -- Executing [s@check_callerid:8] GotoIf("SIP/Toplink-0000002b", "0?9:10") in new stack
    -- Goto (check_callerid,s,10)
    -- Executing [s@check_callerid:10] NoOp("SIP/Toplink-0000002b", "Finish if_for_if_check_callerid_413_414_415") in new stack
    -- Executing [s@check_callerid:11] MSet("SIP/Toplink-0000002b", "LOCAL(x)=7") in new stack
    -- Executing [s@check_callerid:12] Goto("SIP/Toplink-0000002b", "6") in new stack
    -- Goto (check_callerid,s,6)
    -- Executing [s@check_callerid:6] GotoIf("SIP/Toplink-0000002b", "1?7:13") in new stack
    -- Goto (check_callerid,s,7)
    -- Executing [s@check_callerid:7] Set("SIP/Toplink-0000002b", "LOCAL(Test)=0049152") in new stack
    -- Executing [s@check_callerid:8] GotoIf("SIP/Toplink-0000002b", "0?9:10") in new stack
    -- Goto (check_callerid,s,10)
    -- Executing [s@check_callerid:10] NoOp("SIP/Toplink-0000002b", "Finish if_for_if_check_callerid_413_414_415") in new stack
    -- Executing [s@check_callerid:11] MSet("SIP/Toplink-0000002b", "LOCAL(x)=6") in new stack
    -- Executing [s@check_callerid:12] Goto("SIP/Toplink-0000002b", "6") in new stack
    -- Goto (check_callerid,s,6)
    -- Executing [s@check_callerid:6] GotoIf("SIP/Toplink-0000002b", "1?7:13") in new stack
    -- Goto (check_callerid,s,7)
    -- Executing [s@check_callerid:7] Set("SIP/Toplink-0000002b", "LOCAL(Test)=004915") in new stack
    -- Executing [s@check_callerid:8] GotoIf("SIP/Toplink-0000002b", "0?9:10") in new stack
    -- Goto (check_callerid,s,10)
    -- Executing [s@check_callerid:10] NoOp("SIP/Toplink-0000002b", "Finish if_for_if_check_callerid_413_414_415") in new stack
    -- Executing [s@check_callerid:11] MSet("SIP/Toplink-0000002b", "LOCAL(x)=5") in new stack
    -- Executing [s@check_callerid:12] Goto("SIP/Toplink-0000002b", "6") in new stack
    -- Goto (check_callerid,s,6)
    -- Executing [s@check_callerid:6] GotoIf("SIP/Toplink-0000002b", "1?7:13") in new stack
    -- Goto (check_callerid,s,7)
    -- Executing [s@check_callerid:7] Set("SIP/Toplink-0000002b", "LOCAL(Test)=00491") in new stack
    -- Executing [s@check_callerid:8] GotoIf("SIP/Toplink-0000002b", "0?9:10") in new stack
    -- Goto (check_callerid,s,10)
    -- Executing [s@check_callerid:10] NoOp("SIP/Toplink-0000002b", "Finish if_for_if_check_callerid_413_414_415") in new stack
    -- Executing [s@check_callerid:11] MSet("SIP/Toplink-0000002b", "LOCAL(x)=4") in new stack
    -- Executing [s@check_callerid:12] Goto("SIP/Toplink-0000002b", "6") in new stack
    -- Goto (check_callerid,s,6)
    -- Executing [s@check_callerid:6] GotoIf("SIP/Toplink-0000002b", "1?7:13") in new stack
    -- Goto (check_callerid,s,7)
    -- Executing [s@check_callerid:7] Set("SIP/Toplink-0000002b", "LOCAL(Test)=0049") in new stack
    -- Executing [s@check_callerid:8] GotoIf("SIP/Toplink-0000002b", "0?9:10") in new stack
    -- Goto (check_callerid,s,10)
    -- Executing [s@check_callerid:10] NoOp("SIP/Toplink-0000002b", "Finish if_for_if_check_callerid_413_414_415") in new stack
    -- Executing [s@check_callerid:11] MSet("SIP/Toplink-0000002b", "LOCAL(x)=3") in new stack
    -- Executing [s@check_callerid:12] Goto("SIP/Toplink-0000002b", "6") in new stack
    -- Goto (check_callerid,s,6)
    -- Executing [s@check_callerid:6] GotoIf("SIP/Toplink-0000002b", "1?7:13") in new stack
    -- Goto (check_callerid,s,7)
    -- Executing [s@check_callerid:7] Set("SIP/Toplink-0000002b", "LOCAL(Test)=004") in new stack
    -- Executing [s@check_callerid:8] GotoIf("SIP/Toplink-0000002b", "0?9:10") in new stack
    -- Goto (check_callerid,s,10)
    -- Executing [s@check_callerid:10] NoOp("SIP/Toplink-0000002b", "Finish if_for_if_check_callerid_413_414_415") in new stack
    -- Executing [s@check_callerid:11] MSet("SIP/Toplink-0000002b", "LOCAL(x)=2") in new stack
    -- Executing [s@check_callerid:12] Goto("SIP/Toplink-0000002b", "6") in new stack
    -- Goto (check_callerid,s,6)
    -- Executing [s@check_callerid:6] GotoIf("SIP/Toplink-0000002b", "0?7:13") in new stack
    -- Goto (check_callerid,s,13)
    -- Executing [s@check_callerid:13] NoOp("SIP/Toplink-0000002b", "Finish for_if_check_callerid_413_414") in new stack
    -- Executing [s@check_callerid:14] Return("SIP/Toplink-0000002b", "1") in new stack
    -- Executing [s@handle_inbound:13] GotoIf("SIP/Toplink-0000002b", "1?14:15") in new stack
    -- Goto (handle_inbound,s,14)
    -- Executing [s@handle_inbound:14] Goto("SIP/Toplink-0000002b", "handle_inbound,1,1") in new stack
    -- Goto (handle_inbound,1,1)
    -- Executing [1@handle_inbound:1] GotoIf("SIP/Toplink-0000002b", "1?2:5") in new stack
    -- Goto (handle_inbound,1,2)
    -- Executing [1@handle_inbound:2] GotoIf("SIP/Toplink-0000002b", "1?3:4") in new stack
    -- Goto (handle_inbound,1,3)
    -- Executing [1@handle_inbound:4] NoOp("SIP/Toplink-0000002b", "Finish if_if_handle_inbound_424_425") in new stack
    -- Executing [1@handle_inbound:5] NoOp("SIP/Toplink-0000002b", "Finish if_handle_inbound_424") in new stack
    -- Executing [1@handle_inbound:6] Set("SIP/Toplink-0000002b", "CDR(CFD)=") in new stack
[2017-11-14 13:40:04] WARNING[1393]: func_cdr.c:383 cdr_write_callback: CDR requires a value (CDR(variable)=value)
    -- Executing [1@handle_inbound:7] GotoIf("SIP/Toplink-0000002b", "0?8:29") in new stack
    -- Goto (handle_inbound,1,29)
    -- Executing [1@handle_inbound:29] GotoIf("SIP/Toplink-0000002b", "0?30:35") in new stack
    -- Goto (handle_inbound,1,35)
    -- Executing [1@handle_inbound:35] Set("SIP/Toplink-0000002b", "dialtime=90") in new stack
    -- Executing [1@handle_inbound:36] NoOp("SIP/Toplink-0000002b", "Finish if_if_handle_inbound_426_430") in new stack
    -- Executing [1@handle_inbound:37] GotoIf("SIP/Toplink-0000002b", "1?38:55") in new stack
    -- Goto (handle_inbound,1,38)
    -- Executing [1@handle_inbound:38] GotoIf("SIP/Toplink-0000002b", "0?39:51") in new stack
    -- Goto (handle_inbound,1,51)
    -- Executing [1@handle_inbound:51] Set("SIP/Toplink-0000002b", "nst=300") in new stack
  == Begin MixMonitor Recording SIP/Toplink-0000002b
    -- Executing [1@handle_inbound:52] Goto("SIP/Toplink-0000002b", "handle_inbound,1,1") in new stack
    -- Goto (handle_inbound,1,1)
    -- Executing [1@handle_inbound:1] GotoIf("SIP/Toplink-0000002b", "0?2:5") in new stack
    -- Goto (handle_inbound,1,5)
    -- Executing [1@handle_inbound:5] NoOp("SIP/Toplink-0000002b", "Finish if_handle_inbound_424") in new stack
    -- Executing [1@handle_inbound:6] Set("SIP/Toplink-0000002b", "CDR(CFD)=") in new stack
[2017-11-14 13:40:04] WARNING[1393]: func_cdr.c:383 cdr_write_callback: CDR requires a value (CDR(variable)=value)
    -- Executing [1@handle_inbound:7] GotoIf("SIP/Toplink-0000002b", "0?8:29") in new stack
    -- Goto (handle_inbound,1,29)
    -- Executing [1@handle_inbound:29] GotoIf("SIP/Toplink-0000002b", "0?30:35") in new stack
    -- Goto (handle_inbound,1,35)
    -- Executing [1@handle_inbound:35] Set("SIP/Toplink-0000002b", "dialtime=90") in new stack
    -- Executing [1@handle_inbound:36] NoOp("SIP/Toplink-0000002b", "Finish if_if_handle_inbound_426_430") in new stack
    -- Executing [1@handle_inbound:37] GotoIf("SIP/Toplink-0000002b", "1?38:55") in new stack
    -- Goto (handle_inbound,1,38)
    -- Executing [1@handle_inbound:38] GotoIf("SIP/Toplink-0000002b", "1?39:51") in new stack
    -- Goto (handle_inbound,1,39)
    -- Executing [1@handle_inbound:39] Set("SIP/Toplink-0000002b", "GROUP(300)=300") in new stack
    -- Executing [1@handle_inbound:40] GotoIf("SIP/Toplink-0000002b", "1?41:48") in new stack
    -- Goto (handle_inbound,1,41)
    -- Executing [1@handle_inbound:41] Gosub("SIP/Toplink-0000002b", "check_dialstring,s,1") in new stack
    -- Executing [s@check_dialstring:1] Set("SIP/Toplink-0000002b", "LOCAL(tmp)=SIP/300") in new stack
    -- Executing [s@check_dialstring:2] Set("SIP/Toplink-0000002b", "ToDial=") in new stack
    -- Executing [s@check_dialstring:3] MSet("SIP/Toplink-0000002b", "i=1") in new stack
    -- Executing [s@check_dialstring:4] GotoIf("SIP/Toplink-0000002b", "1?5:11") in new stack
    -- Goto (check_dialstring,s,5)
    -- Executing [s@check_dialstring:5] Set("SIP/Toplink-0000002b", "LOCAL(nsttmp)=SIP/300") in new stack
    -- Executing [s@check_dialstring:6] GotoIf("SIP/Toplink-0000002b", "1?7:8") in new stack
    -- Goto (check_dialstring,s,7)
    -- Executing [s@check_dialstring:7] Set("SIP/Toplink-0000002b", "ToDial=&SIP/300") in new stack
    -- Executing [s@check_dialstring:8] NoOp("SIP/Toplink-0000002b", "Finish if_for_check_dialstring_416_417") in new stack
    -- Executing [s@check_dialstring:9] MSet("SIP/Toplink-0000002b", "i=2") in new stack
    -- Executing [s@check_dialstring:10] Goto("SIP/Toplink-0000002b", "4") in new stack
    -- Goto (check_dialstring,s,4)
    -- Executing [s@check_dialstring:4] GotoIf("SIP/Toplink-0000002b", "0?5:11") in new stack
    -- Goto (check_dialstring,s,11)
    -- Executing [s@check_dialstring:11] NoOp("SIP/Toplink-0000002b", "Finish for_check_dialstring_416") in new stack
    -- Executing [s@check_dialstring:12] GotoIf("SIP/Toplink-0000002b", "1?13:14") in new stack
    -- Goto (check_dialstring,s,13)
    -- Executing [s@check_dialstring:13] Set("SIP/Toplink-0000002b", "ToDial=SIP/300") in new stack
    -- Executing [s@check_dialstring:14] NoOp("SIP/Toplink-0000002b", "Finish if_check_dialstring_418") in new stack
    -- Executing [s@check_dialstring:15] Return("SIP/Toplink-0000002b", "") in new stack
    -- Executing [1@handle_inbound:42] GotoIf("SIP/Toplink-0000002b", "1?43:45") in new stack
    -- Goto (handle_inbound,1,43)
    -- Executing [1@handle_inbound:43] Dial("SIP/Toplink-0000002b", "SIP/300,90,twhrg") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 11792
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 87.5.162.25:5060:
INVITE sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248 SIP/2.0
Via: SIP/2.0/UDP 145.45.10.63:5060;branch=z9hG4bK56a1b879;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as3a4ef345
To: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.13.1
Date: Tue, 14 Nov 2017 12:40:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Type: application/sdp
Content-Length: 274

v=0
o=root 1199224174 1199224174 IN IP4 2.5.21.85
s=Asterisk PBX 13.13.1
c=IN IP4 2.5.21.85
t=0 0
m=audio 11792 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Called SIP/300

<--- SIP read from UDP:87.147.162.207:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 2.5.21.85:5060;branch=z9hG4bK56a1b879;rport=5060
From: <sip:[email protected]>;tag=as3a4ef345
To: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: AVM FRITZ!Box 7490 113.06.92 (Oct 12 2017)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:87.147.162.207:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 2.5.21.85:5060;branch=z9hG4bK56a1b879;rport=5060
From: <sip:[email protected]>;tag=as3a4ef345
To: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>;tag=27C6E9FA5FC1D498
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>
User-Agent: AVM FRITZ!Box 7490 113.06.92 (Oct 12 2017)
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>
    -- SIP/300-0000002c is ringing

<--- SIP read from UDP:87.147.162.207:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.5.21.85:5060;branch=z9hG4bK56a1b879;rport=5060
From: <sip:[email protected]>;tag=as3a4ef345
To: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>;tag=27C6E9FA5FC1D498
Call-ID: [email protected]
CSeq: 102 INVITE
Contact: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>
User-Agent: AVM FRITZ!Box 7490 113.06.92 (Oct 12 2017)
Supported: 100rel,replaces
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH
Content-Type: application/sdp
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 272

v=0
o=user 16436981 16436981 IN IP4 87.147.162.207
s=Asterisk PBX 13.13.1
c=IN IP4 87.147.162.207
t=0 0
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:7079
a=ptime:20
<------------->
--- (15 headers 13 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 87.147.162.207:7078
sip_route_dump: route/path hop: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>
Transmitting (NAT) to 87.147.162.207:5060:
ACK sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248 SIP/2.0
Via: SIP/2.0/UDP 2.5.21.85:5060;branch=z9hG4bK3839f6d9;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as3a4ef345
To: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>;tag=27C6E9FA5FC1D498
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.13.1
Content-Length: 0


---
    -- SIP/300-0000002c answered SIP/Toplink-0000002b
    -- Channel SIP/300-0000002c joined 'simple_bridge' basic-bridge <20307977-f674-4ac5-9226-26ed5c839bb0>
    -- Channel SIP/Toplink-0000002b joined 'simple_bridge' basic-bridge <20307977-f674-4ac5-9226-26ed5c839bb0>
       > 0x7f669c00fc80 -- Probation passed - setting RTP source address to 87.147.162.207:7078
       > 0x7f66b03220c0 -- Probation passed - setting RTP source address to 213.218.12.40:10450
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
Reliably Transmitting (NAT) to 87.147.162.207:5060:
INVITE sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248 SIP/2.0
Via: SIP/2.0/UDP 2.5.21.85:5060;branch=z9hG4bK23803e35;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as3a4ef345
To: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>;tag=27C6E9FA5FC1D498
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX 13.13.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 1199224174 1199224175 IN IP4 2.5.21.85
s=Asterisk PBX 13.13.1
c=IN IP4 2.5.21.85
t=0 0
m=image 4183 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:284
a=T38FaxUdpEC:t38UDPFEC

---
    -- Started music on hold, class 'default', on channel 'SIP/300-0000002c'
       > 0x7f66b03220c0 -- Probation passed - setting RTP source address to 213.218.12.40:10450
[2017-11-14 13:40:09] WARNING[6261][C-00000044]: mp3/interface.c:217 decodeMP3: Junk at the beginning of frame 49443304

<--- SIP read from UDP:87.147.162.207:5060 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 2.5.21.85:5060;branch=z9hG4bK23803e35;rport=5060
From: <sip:[email protected]>;tag=as3a4ef345
To: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>;tag=27C6E9FA5FC1D498
Call-ID: [email protected]
CSeq: 103 INVITE
Warning: 399 0.0.0.0 "successful but result empty"
User-Agent: FRITZ!OS
Content-Type: application/sdp
Content-Length: 375

v=0
o=user 16436981 16436982 IN IP4 87.147.162.207
s=call
c=IN IP4 87.147.162.207
t=0 0
m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101
a=sendrecv
a=rtpmap:2 G726-32/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:7079
a=ptime:20
<------------->
--- (10 headers 17 lines) ---
Transmitting (NAT) to 87.147.162.207:5060:
ACK sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248 SIP/2.0
Via: SIP/2.0/UDP 2.5.21.85:5060;branch=z9hG4bK23803e35;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as3a4ef345
To: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>;tag=27C6E9FA5FC1D498
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX 13.13.1
Content-Length: 0


---
Audio is at 11792
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 87.147.162.207:5060:
INVITE sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248 SIP/2.0
Via: SIP/2.0/UDP 2.5.21.85:5060;branch=z9hG4bK4d6e34d1;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as3a4ef345
To: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>;tag=27C6E9FA5FC1D498
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 104 INVITE
User-Agent: Asterisk PBX 13.13.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 274

v=0
o=root 1199224174 1199224176 IN IP4 2.5.21.85
s=Asterisk PBX 13.13.1
c=IN IP4 2.5.21.85
t=0 0
m=audio 11792 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
       > 0x7f66b03220c0 -- Probation passed - setting RTP source address to 213.218.12.40:10450

<--- SIP read from UDP:87.147.162.207:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.5.21.85:5060;branch=z9hG4bK4d6e34d1;rport=5060
From: <sip:[email protected]>;tag=as3a4ef345
To: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>;tag=27C6E9FA5FC1D498
Call-ID: [email protected]
CSeq: 104 INVITE
Contact: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>
User-Agent: AVM FRITZ!Box 7490 113.06.92 (Oct 12 2017)
Supported: 100rel,replaces
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH
Content-Type: application/sdp
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 272

v=0
o=user 16436981 16436983 IN IP4 87.147.162.207
s=Asterisk PBX 13.13.1
c=IN IP4 87.147.162.207
t=0 0
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:7079
a=ptime:20
<------------->
--- (15 headers 13 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 87.147.162.207:7078
Transmitting (NAT) to 87.147.162.207:5060:
ACK sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248 SIP/2.0
Via: SIP/2.0/UDP 2.5.21.85:5060;branch=z9hG4bK0c834a04;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as3a4ef345
To: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>;tag=27C6E9FA5FC1D498
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 104 ACK
User-Agent: Asterisk PBX 13.13.1
Content-Length: 0


---
Reliably Transmitting (NAT) to 87.147.162.207:5060:
INVITE sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248 SIP/2.0
Via: SIP/2.0/UDP 2.5.21.85:5060;branch=z9hG4bK35159230;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as3a4ef345
To: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>;tag=27C6E9FA5FC1D498
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 105 INVITE
User-Agent: Asterisk PBX 13.13.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 1199224174 1199224177 IN IP4 2.5.21.85
s=Asterisk PBX 13.13.1
c=IN IP4 2.5.21.85
t=0 0
m=image 4183 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:284
a=T38FaxUdpEC:t38UDPFEC

---
    -- Stopped music on hold on SIP/300-0000002c
       > 0x7f669c00fc80 -- Probation passed - setting RTP source address to 87.147.162.207:7078
       > 0x7f66b03220c0 -- Probation passed - setting RTP source address to 213.218.12.40:10450
    -- Channel SIP/Toplink-0000002b left 'simple_bridge' basic-bridge <20307977-f674-4ac5-9226-26ed5c839bb0>
  == Spawn extension (handle_inbound, 1, 43) exited non-zero on 'SIP/Toplink-0000002b'
    -- Channel SIP/300-0000002c left 'simple_bridge' basic-bridge <20307977-f674-4ac5-9226-26ed5c839bb0>
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
    -- Executing [h@handle_inbound:1] System("SIP/Toplink-0000002b", "/etc/asterisk/consolidatecdr.sh v22017024287244466-1510663203.45 "ANSWER" &") in new stack

<--- SIP read from UDP:87.147.162.207:5060 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP 2.5.21.85:5060;branch=z9hG4bK35159230;rport=5060
From: <sip:[email protected]>;tag=as3a4ef345
To: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>;tag=27C6E9FA5FC1D498
Call-ID: [email protected]
CSeq: 105 INVITE
Warning: 399 0.0.0.0 "successful but result empty"
User-Agent: FRITZ!OS
Content-Type: application/sdp
Content-Length: 375

v=0
o=user 16436981 16436984 IN IP4 87.147.162.207
s=call
c=IN IP4 87.147.162.207
t=0 0
m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101
a=sendrecv
a=rtpmap:2 G726-32/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:7079
a=ptime:20
<------------->
--- (10 headers 17 lines) ---
Transmitting (NAT) to 87.147.162.207:5060:
ACK sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248 SIP/2.0
Via: SIP/2.0/UDP 2.5.21.85:5060;branch=z9hG4bK35159230;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as3a4ef345
To: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>;tag=27C6E9FA5FC1D498
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 105 ACK
User-Agent: Asterisk PBX 13.13.1
Content-Length: 0


---
Audio is at 11792
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 87.147.162.207:5060:
INVITE sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248 SIP/2.0
Via: SIP/2.0/UDP 2.5.21.85:5060;branch=z9hG4bK62383f9a;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as3a4ef345
To: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>;tag=27C6E9FA5FC1D498
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 106 INVITE
User-Agent: Asterisk PBX 13.13.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 301

v=0
o=root 1199224174 1199224178 IN IP4 2.5.21.85
s=Asterisk PBX 13.13.1
c=IN IP4 2.5.21.85
t=0 0
m=audio 11792 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Executing [h@handle_inbound:2] GotoIf("SIP/Toplink-0000002b", "1?3:5") in new stack
    -- Goto (handle_inbound,h,3)
    -- Executing [h@handle_inbound:3] StopMixMonitor("SIP/Toplink-0000002b", "") in new stack
  == MixMonitor close filestream (mixed)
    -- Executing [h@handle_inbound:4] Set("SIP/Toplink-0000002b", "__HASMONITOR=") in new stack
    -- Executing [h@handle_inbound:5] NoOp("SIP/Toplink-0000002b", "Finish if_handle_inbound_453") in new stack
    -- Executing [h@handle_inbound:6] Hangup("SIP/Toplink-0000002b", "") in new stack
  == Spawn extension (handle_inbound, h, 6) exited non-zero on 'SIP/Toplink-0000002b'
  == Executing [/etc/asterisk/wav2mp3.sh v22017024287244466-1510663203.45-XXXXXXXXXX-300-20171114-134004.wav]

<--- SIP read from UDP:87.147.162.207:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.5.21.85:5060;branch=z9hG4bK62383f9a;rport=5060
From: <sip:[email protected]>;tag=as3a4ef345
To: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>;tag=27C6E9FA5FC1D498
Call-ID: [email protected]
CSeq: 106 INVITE
Contact: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>
User-Agent: AVM FRITZ!Box 7490 113.06.92 (Oct 12 2017)
Supported: 100rel,replaces
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH
Content-Type: application/sdp
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 272

v=0
o=user 16436981 16436985 IN IP4 87.147.162.207
s=Asterisk PBX 13.13.1
c=IN IP4 87.147.162.207
t=0 0
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:7079
a=ptime:20
<------------->
--- (15 headers 13 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 87.147.162.207:7078
Transmitting (NAT) to 87.147.162.207:5060:
ACK sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248 SIP/2.0
Via: SIP/2.0/UDP 2.5.21.85:5060;branch=z9hG4bK7cf3e18f;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as3a4ef345
To: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>;tag=27C6E9FA5FC1D498
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 106 ACK
User-Agent: Asterisk PBX 13.13.1
Content-Length: 0


---
Reliably Transmitting (NAT) to 87.147.162.207:5060:
BYE sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248 SIP/2.0
Via: SIP/2.0/UDP 2.5.21.85:5060;branch=z9hG4bK6782c838;rport
Max-Forwards: 70
From: <sip:[email protected]>;tag=as3a4ef345
To: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>;tag=27C6E9FA5FC1D498
Call-ID: [email protected]
CSeq: 107 BYE
User-Agent: Asterisk PBX 13.13.1
Reason: Q.850;cause=16
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
  == End MixMonitor Recording SIP/Toplink-0000002b

<--- SIP read from UDP:87.147.162.207:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 2.5.21.85:5060;branch=z9hG4bK6782c838;rport=5060
From: <sip:[email protected]>;tag=as3a4ef345
To: <sip:[email protected];uniq=2AA2898BCFE0A540E40236ABED248>;tag=27C6E9FA5FC1D498
Call-ID: [email protected]
CSeq: 107 BYE
X-RTP-Stat: CS=0;PS=0;ES=12;OS=0;SP=0/0;SO=0;QS=-;PR=7;ER=12;OR=1120;CR=0;SR=0;QR=-;PL=0,0;BL=0;LS=0;RB=0/0;SB=-/-;EN=PCMU;DE=PCMU;JI=5,0;DL=0,0,0;IP=87.147.162.207:7078,2.5.21.85:11792
X-RTP-Stat-Add: DQ=0;DSS=0;DS=0;PLCS=0;JS=0
X-SIP-Stat: DRT=0;IR=0
User-Agent: AVM FRITZ!Box 7490 113.06.92 (Oct 12 2017)
Supported: 100rel,replaces
Allow-Events: telephone-event,refer
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: INVITE
v22017024287244466*CLI> exit
Asterisk cleanly ending (0).
Executing last minute cleanups
 
Zuletzt bearbeitet von einem Moderator:
Um antworten zu können musst du eingeloggt sein.
Kostenlos!
Jetzt holen

Neueste Beiträge

Kategorien

Wissenswertes

Statistik des Forums

Themen
248,854
Beiträge
2,302,940
Mitglieder
378,504
Neuestes Mitglied
Hallo61Hallo

3CX jetzt einen Monat kostenlos testen

Die Verbindung, Zusammenarbeit und Kommunikation mit Ihrem Team und Ihren Kunden war noch nie so einfach. Nutzen Sie unsere kostenlose 30-Tage-Testversion und entdecken Sie die richtige Lösung für Ihr Unternehmen.

3CX start your free trial guy
Für diese E-Mail-Adresse besteht bereits ein 3CX-Konto. Sie werden zum Kundenportal weitergeleitet, wo Sie sich anmelden oder Ihr Passwort zurücksetzen können, falls Sie dieses vergessen haben.
| Style by ThemeHouse
XenPorta 2 PRO © Jason Axelrod of 8WAYRUN
Oben Unten