Heute hatte ich die Zeit/Gelegenheit - mal Asterisk im Debug-Modus laufen zu lassen und dabei Calls auszuführen.
Raus gekommen sind leider, nicht wirklich aussagekräftige Informationen - ich will sie aber trotzdem posten:
Wer hat noch Hinweise für weitere Debugging-Varianten mit Asterisk?
Interessant ist, das das h264-Codec immer angeboten, teilweise sogar angenommen wird.
[table="width: 500, class: grid, align: left"]
[tr]
[td][/td]
[td]6000-6004[/td]
[td]6000-6007[/td]
[td]6007-6003[/td]
[/tr]
[tr]
[td]call[/td]
[td][Oct 10 11:06:07] VERBOSE[918] netsock.c: == Using SIP RTP CoS mark 5
[Oct 10 11:06:07] VERBOSE[918] netsock.c: == Using SIP VRTP CoS mark 6
[Oct 10 11:06:07] DEBUG[918] chan_sip.c: Setting NAT on RTP to Off
[Oct 10 11:06:07] DEBUG[918] chan_sip.c: Setting NAT on VRTP to Off
[Oct 10 11:06:07] DEBUG[918] chan_sip.c: Allocating new SIP dialog for
[email protected] - INVITE (With RTP)
[Oct 10 11:06:07] DEBUG[918] chan_sip.c: Setting NAT on RTP to On
[Oct 10 11:06:07] DEBUG[918] chan_sip.c: Setting NAT on VRTP to On
[Oct 10 11:06:07] DEBUG[918] chan_sip.c: Stopping retransmission on '
[email protected]' of Response 1: Match Found
[Oct 10 11:06:07] DEBUG[918] chan_sip.c: Setting NAT on RTP to On
[Oct 10 11:06:07] DEBUG[918] chan_sip.c: Setting NAT on VRTP to On
[Oct 10 11:06:07] DEBUG[918] chan_sip.c: Checking SIP call limits for device 6000
[Oct 10 11:06:07] DEBUG[1010] pbx.c: Launching 'Dial'
[Oct 10 11:06:07] VERBOSE[1010] pbx.c: -- Executing [6004@DLPN_DialPlan1:1] Dial("SIP/6000-00000000", "SIP/6004") in new stack
[Oct 10 11:06:07] DEBUG[1010] app_dial.c: chan->name=SIP/6000-00000000, transfercapability=0x0.
[Oct 10 11:06:07] DEBUG[1010] channel.c: type=SIP, format=0x4
[Oct 10 11:06:07] DEBUG[1010] channel.c: fmt=0x4, capabilities=0xffff.
[Oct 10 11:06:07] DEBUG[1010] channel.c: capabilities=0x4, videoformat=0x0, textformat=0x0.
[Oct 10 11:06:07] DEBUG[1010] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw)
[Oct 10 11:06:07] VERBOSE[1010] netsock.c: == Using SIP RTP CoS mark 5
[Oct 10 11:06:07] VERBOSE[1010] netsock.c: == Using SIP VRTP CoS mark 6
[Oct 10 11:06:07] DEBUG[1010] chan_sip.c: Allocating new SIP dialog for
[email protected] - INVITE (With RTP)
[Oct 10 11:06:07] DEBUG[1010] chan_sip.c: Setting NAT on RTP to On
[Oct 10 11:06:07] DEBUG[1010] chan_sip.c: Setting NAT on VRTP to On
[Oct 10 11:06:07] DEBUG[1010] channel.c: Not copying variable DIALEDTIME.
[Oct 10 11:06:07] DEBUG[1010] channel.c: Not copying variable ANSWEREDTIME.
[Oct 10 11:06:07] DEBUG[1010] channel.c: Not copying variable DIALEDPEERNAME.
[Oct 10 11:06:07] DEBUG[1010] channel.c: Not copying variable DIALEDPEERNUMBER.
[Oct 10 11:06:07] DEBUG[1010] channel.c: Not copying variable DIALSTATUS.
[Oct 10 11:06:07] DEBUG[1010] channel.c: Not copying variable SIPCALLID.
[Oct 10 11:06:07] DEBUG[1010] channel.c: Not copying variable SIPDOMAIN.
[Oct 10 11:06:07] DEBUG[1010] channel.c: Not copying variable SIPURI.
[Oct 10 11:06:07] DEBUG[1010] chan_sip.c: Outgoing Call for 6004
[Oct 10 11:06:07] DEBUG[1010] chan_sip.c: ** Our capability: 0x200104 (ulaw|g729|h264) Video flag: False Text flag: False
[Oct 10 11:06:07] DEBUG[1010] chan_sip.c: ** Our prefcodec: 0x4 (ulaw)
[Oct 10 11:06:07] DEBUG[1010] chan_sip.c: Initializing initreq for method INVITE - callid
[email protected]
[Oct 10 11:06:07] VERBOSE[1010] app_dial.c: -- Called 6004
[Oct 10 11:06:07] DEBUG[918] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '
[email protected]' Request 102: Found
[Oct 10 11:06:07] DEBUG[918] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '
[email protected]' Request 102: Found
[Oct 10 11:06:07] VERBOSE[1010] app_dial.c: -- SIP/6004-00000001 is ringing[/td]
[td][Oct 10 11:07:53] VERBOSE[918] netsock.c: == Using SIP RTP CoS mark 5
[Oct 10 11:07:53] VERBOSE[918] netsock.c: == Using SIP VRTP CoS mark 6
[Oct 10 11:07:53] DEBUG[918] chan_sip.c: Setting NAT on RTP to Off
[Oct 10 11:07:53] DEBUG[918] chan_sip.c: Setting NAT on VRTP to Off
[Oct 10 11:07:53] DEBUG[918] chan_sip.c: Allocating new SIP dialog for
[email protected] - INVITE (With RTP)
[Oct 10 11:07:53] DEBUG[918] chan_sip.c: Setting NAT on RTP to On
[Oct 10 11:07:53] DEBUG[918] chan_sip.c: Setting NAT on VRTP to On
[Oct 10 11:07:53] DEBUG[918] chan_sip.c: Stopping retransmission on '
[email protected]' of Response 1: Match Found
[Oct 10 11:07:53] DEBUG[918] chan_sip.c: Setting NAT on RTP to On
[Oct 10 11:07:53] DEBUG[918] chan_sip.c: Setting NAT on VRTP to On
[Oct 10 11:07:53] DEBUG[918] chan_sip.c: Checking SIP call limits for device 6000
[Oct 10 11:07:53] DEBUG[1015] pbx.c: Launching 'Dial'
[Oct 10 11:07:53] VERBOSE[1015] pbx.c: -- Executing [6007@DLPN_DialPlan1:1] Dial("SIP/6000-00000002", "SIP/6007") in new stack
[Oct 10 11:07:53] DEBUG[1015] app_dial.c: chan->name=SIP/6000-00000002, transfercapability=0x0.
[Oct 10 11:07:53] DEBUG[1015] channel.c: type=SIP, format=0x4
[Oct 10 11:07:53] DEBUG[1015] channel.c: fmt=0x4, capabilities=0xffff.
[Oct 10 11:07:53] DEBUG[1015] channel.c: capabilities=0x4, videoformat=0x0, textformat=0x0.
[Oct 10 11:07:53] DEBUG[1015] chan_sip.c: Asked to create a SIP channel with formats: 0x4 (ulaw)
[Oct 10 11:07:53] VERBOSE[1015] netsock.c: == Using SIP RTP CoS mark 5
[Oct 10 11:07:53] VERBOSE[1015] netsock.c: == Using SIP VRTP CoS mark 6
[Oct 10 11:07:53] DEBUG[1015] chan_sip.c: Allocating new SIP dialog for
[email protected] - INVITE (With RTP)
[Oct 10 11:07:53] DEBUG[1015] chan_sip.c: Setting NAT on RTP to On
[Oct 10 11:07:53] DEBUG[1015] chan_sip.c: Setting NAT on VRTP to On
[Oct 10 11:07:53] DEBUG[1015] channel.c: Not copying variable DIALEDTIME.
[Oct 10 11:07:53] DEBUG[1015] channel.c: Not copying variable ANSWEREDTIME.
[Oct 10 11:07:53] DEBUG[1015] channel.c: Not copying variable DIALEDPEERNAME.
[Oct 10 11:07:53] DEBUG[1015] channel.c: Not copying variable DIALEDPEERNUMBER.
[Oct 10 11:07:53] DEBUG[1015] channel.c: Not copying variable DIALSTATUS.
[Oct 10 11:07:53] DEBUG[1015] channel.c: Not copying variable SIPCALLID.
[Oct 10 11:07:53] DEBUG[1015] channel.c: Not copying variable SIPDOMAIN.
[Oct 10 11:07:53] DEBUG[1015] channel.c: Not copying variable SIPURI.
[Oct 10 11:07:53] DEBUG[1015] chan_sip.c: Outgoing Call for 6007
[Oct 10 11:07:53] DEBUG[1015] chan_sip.c: ** Our capability: 0x200004 (ulaw|h264) Video flag: False Text flag: False
[Oct 10 11:07:53] DEBUG[1015] chan_sip.c: ** Our prefcodec: 0x4 (ulaw)
[Oct 10 11:07:53] DEBUG[1015] chan_sip.c: Initializing initreq for method INVITE - callid
[email protected]
[Oct 10 11:07:53] VERBOSE[1015] app_dial.c: -- Called 6007
[Oct 10 11:07:54] DEBUG[918] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '
[email protected]' Request 102: Found
[Oct 10 11:07:54] DEBUG[918] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '
[email protected]' Request 102: Found
[Oct 10 11:07:54] VERBOSE[1015] app_dial.c: -- SIP/6007-00000003 is ringing[/td]
[td][Oct 10 11:11:27] VERBOSE[918] netsock.c: == Using SIP RTP CoS mark 5
[Oct 10 11:11:27] VERBOSE[918] netsock.c: == Using SIP VRTP CoS mark 6
[Oct 10 11:11:27] DEBUG[918] chan_sip.c: Setting NAT on RTP to Off
[Oct 10 11:11:27] DEBUG[918] chan_sip.c: Setting NAT on VRTP to Off
[Oct 10 11:11:27] DEBUG[918] chan_sip.c: Allocating new SIP dialog for
[email protected] - INVITE (With RTP)
[Oct 10 11:11:27] DEBUG[918] chan_sip.c: Setting NAT on RTP to On
[Oct 10 11:11:27] DEBUG[918] chan_sip.c: Setting NAT on VRTP to On
[Oct 10 11:11:27] DEBUG[918] chan_sip.c: Stopping retransmission on '
[email protected]' of Response 101: Match Found
[Oct 10 11:11:27] DEBUG[918] chan_sip.c: Setting NAT on RTP to On
[Oct 10 11:11:27] DEBUG[918] chan_sip.c: Setting NAT on VRTP to On
[Oct 10 11:11:27] DEBUG[918] chan_sip.c: Checking SIP call limits for device 6007
[Oct 10 11:11:27] DEBUG[1024] pbx.c: Launching 'Dial'
[Oct 10 11:11:27] VERBOSE[1024] pbx.c: -- Executing [6003@DLPN_DialPlan1:1] Dial("SIP/6007-00000006", "SIP/6003") in new stack
[Oct 10 11:11:27] DEBUG[1024] app_dial.c: chan->name=SIP/6007-00000006, transfercapability=0x0.
[Oct 10 11:11:27] DEBUG[1024] channel.c: type=SIP, format=0x200004
[Oct 10 11:11:27] DEBUG[1024] channel.c: fmt=0x4, capabilities=0xffff.
[Oct 10 11:11:27] DEBUG[1024] channel.c: capabilities=0x4, videoformat=0x200000, textformat=0x0.
[Oct 10 11:11:27] DEBUG[1024] chan_sip.c: Asked to create a SIP channel with formats: 0x200004 (ulaw|h264)
[Oct 10 11:11:27] VERBOSE[1024] netsock.c: == Using SIP RTP CoS mark 5
[Oct 10 11:11:27] VERBOSE[1024] netsock.c: == Using SIP VRTP CoS mark 6
[Oct 10 11:11:27] DEBUG[1024] chan_sip.c: Allocating new SIP dialog for
[email protected] - INVITE (With RTP)
[Oct 10 11:11:27] DEBUG[1024] chan_sip.c: Setting NAT on RTP to Off
[Oct 10 11:11:27] DEBUG[1024] chan_sip.c: Setting NAT on VRTP to Off
[Oct 10 11:11:27] DEBUG[1024] channel.c: Not copying variable DIALEDTIME.
[Oct 10 11:11:27] DEBUG[1024] channel.c: Not copying variable ANSWEREDTIME.
[Oct 10 11:11:27] DEBUG[1024] channel.c: Not copying variable DIALEDPEERNAME.
[Oct 10 11:11:27] DEBUG[1024] channel.c: Not copying variable DIALEDPEERNUMBER.
[Oct 10 11:11:27] DEBUG[1024] channel.c: Not copying variable DIALSTATUS.
[Oct 10 11:11:27] DEBUG[1024] channel.c: Not copying variable SIPCALLID.
[Oct 10 11:11:27] DEBUG[1024] channel.c: Not copying variable SIPDOMAIN.
[Oct 10 11:11:27] DEBUG[1024] channel.c: Not copying variable SIPURI.
[Oct 10 11:11:27] DEBUG[1024] chan_sip.c: Outgoing Call for 6003
[Oct 10 11:11:27] DEBUG[1024] chan_sip.c: ** Our capability: 0x20000c (ulaw|alaw|h264) Video flag: False Text flag: False
[Oct 10 11:11:27] DEBUG[1024] chan_sip.c: ** Our prefcodec: 0x200004 (ulaw|h264)
[Oct 10 11:11:27] DEBUG[1024] chan_sip.c: Initializing initreq for method INVITE - callid
[email protected]
[Oct 10 11:11:27] VERBOSE[1024] app_dial.c: -- Called 6003
[Oct 10 11:11:27] DEBUG[918] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '
[email protected]' Request 102: Found
[Oct 10 11:11:27] DEBUG[918] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '
[email protected]' Request 102: Found
[Oct 10 11:11:27] VERBOSE[1024] app_dial.c: -- SIP/6003-00000007 is ringing[/td]
[/tr]
[tr]
[td]pickup[/td]
[td][Oct 10 11:06:28] DEBUG[918] chan_sip.c: Acked pending invite 102
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Stopping retransmission on '
[email protected]' of Request 102: Match Found
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 4 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 2 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 1 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 32 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 32 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 128 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 8 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 4096 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 64 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 64 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 32 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 32 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 256 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 65536 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 262144 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 524288 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 1024 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 1048576 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 2097152 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 8192 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 1048576 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 4194304 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 67108864 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 134217728 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 512 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 2048 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 16 to 20
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Setting framing for 16384 to 20
[Oct 10 11:06:28] VERBOSE[1010] app_dial.c: -- SIP/6004-00000001 answered SIP/6000-00000000
[Oct 10 11:06:28] DEBUG[1010] chan_sip.c: SIP answering channel: SIP/6000-00000000
[Oct 10 11:06:28] DEBUG[1010] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True
[Oct 10 11:06:28] DEBUG[1010] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
[Oct 10 11:06:28] DEBUG[918] chan_sip.c: Stopping retransmission on '
[email protected]' of Response 2: Match Found
[Oct 10 11:06:28] DEBUG[1010] rtp.c: Ooh, format changed from unknown to ulaw
[Oct 10 11:06:28] DEBUG[1010] rtp.c: Created smoother: format: 4 ms: 20 len: 160
[Oct 10 11:06:29] DEBUG[1010] rtp.c: Ooh, format changed from unknown to ulaw
[Oct 10 11:06:29] DEBUG[1010] rtp.c: Created smoother: format: 4 ms: 20 len: 160[/td]
[td][Oct 10 11:08:02] DEBUG[918] chan_sip.c: Acked pending invite 102
[Oct 10 11:08:02] DEBUG[918] chan_sip.c: Stopping retransmission on '
[email protected]' of Request 102: Match Found
[Oct 10 11:08:02] VERBOSE[1015] app_dial.c: -- SIP/6007-00000003 answered SIP/6000-00000002
[Oct 10 11:08:02] DEBUG[1015] chan_sip.c: SIP answering channel: SIP/6000-00000002
[Oct 10 11:08:02] DEBUG[1015] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True
[Oct 10 11:08:02] DEBUG[1015] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
[Oct 10 11:08:02] DEBUG[918] chan_sip.c: ** Our capability: 0x200004 (ulaw|h264) Video flag: False Text flag: True
[Oct 10 11:08:02] DEBUG[918] chan_sip.c: ** Our prefcodec: 0x4 (ulaw)
[Oct 10 11:08:02] DEBUG[918] chan_sip.c: Stopping retransmission on '
[email protected]' of Response 101: Match Found
[Oct 10 11:08:02] DEBUG[918] chan_sip.c: Stopping retransmission on '
[email protected]' of Response 2: Match Found
[Oct 10 11:08:02] DEBUG[1015] rtp.c: Ooh, format changed from unknown to ulaw
[Oct 10 11:08:02] DEBUG[1015] rtp.c: Created smoother: format: 4 ms: 20 len: 160
[Oct 10 11:08:03] DEBUG[1015] rtp.c: Ooh, format changed from unknown to ulaw
[Oct 10 11:08:03] DEBUG[1015] rtp.c: Created smoother: format: 4 ms: 20 len: 160
[Oct 10 11:08:07] DEBUG[1015] rtp.c: Got RTCP report of 64 bytes[/td]
[td][Oct 10 11:11:37] DEBUG[918] chan_sip.c: Acked pending invite 102
[Oct 10 11:11:37] DEBUG[918] chan_sip.c: Stopping retransmission on '
[email protected]' of Request 102: Match Found
[Oct 10 11:11:37] VERBOSE[1024] app_dial.c: -- SIP/6003-00000007 answered SIP/6007-00000006
[Oct 10 11:11:37] DEBUG[1024] chan_sip.c: SIP answering channel: SIP/6007-00000006
[Oct 10 11:11:37] DEBUG[1024] chan_sip.c: ** Our capability: 0x200004 (ulaw|h264) Video flag: False Text flag: True
[Oct 10 11:11:37] DEBUG[1024] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
[Oct 10 11:11:37] DEBUG[918] chan_sip.c: ** Our capability: 0x200004 (ulaw|h264) Video flag: False Text flag: True
[Oct 10 11:11:37] DEBUG[918] chan_sip.c: ** Our prefcodec: 0x200004 (ulaw|h264)
[Oct 10 11:11:37] DEBUG[918] chan_sip.c: Stopping retransmission on '
[email protected]' of Response 102: Match Found
[Oct 10 11:11:37] DEBUG[918] chan_sip.c: Stopping retransmission on '
[email protected]' of Response 101: Match Found
[Oct 10 11:11:37] DEBUG[1024] rtp.c: Ooh, format changed from unknown to ulaw
[Oct 10 11:11:37] DEBUG[1024] rtp.c: Created smoother: format: 4 ms: 20 len: 160
[Oct 10 11:11:38] DEBUG[1024] rtp.c: Ooh, format changed from unknown to ulaw
[Oct 10 11:11:38] DEBUG[1024] rtp.c: Created smoother: format: 4 ms: 20 len: 160
[Oct 10 11:11:39] DEBUG[1024] rtp.c: Got RTCP report of 64 bytes
[Oct 10 11:11:42] DEBUG[1024] rtp.c: Got RTCP report of 64 bytes
[Oct 10 11:11:44] DEBUG[1024] rtp.c: Got RTCP report of 64 bytes[/td]
[/tr]
[tr]
[td]Videotaste drücken[/td]
[td][Oct 10 11:06:51] DEBUG[918] chan_sip.c: ** Our capability: 0x200004 (ulaw|h264) Video flag: False Text flag: True
[Oct 10 11:06:51] DEBUG[918] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
[Oct 10 11:06:51] DEBUG[1010] rtp.c: Ooh, format changed from unknown to h264
[Oct 10 11:06:51] DEBUG[918] chan_sip.c: Stopping retransmission on '
[email protected]' of Response 3: Match Found
[Oct 10 11:06:52] DEBUG[1010] rtp.c: Forcing Marker bit, because SSRC has changed
[Oct 10 11:06:52] DEBUG[1010] rtp.c: Ooh, format changed from unknown to h264
[Oct 10 11:06:53] DEBUG[918] chan_sip.c: Stopping retransmission on '
[email protected]' of Request 103: Match Found[/td]
[td][Oct 10 11:08:12] DEBUG[918] chan_sip.c: ** Our capability: 0x200004 (ulaw|h264) Video flag: False Text flag: True
[Oct 10 11:08:12] DEBUG[918] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
[Oct 10 11:08:12] DEBUG[1015] rtp.c: Got RTCP report of 64 bytes
[Oct 10 11:08:12] DEBUG[918] chan_sip.c: Stopping retransmission on '
[email protected]' of Response 3: Match Found
[Oct 10 11:08:13] DEBUG[1015] rtp.c: Forcing Marker bit, because SSRC has changed
[Oct 10 11:08:13] DEBUG[1015] rtp.c: Ooh, format changed from unknown to h264
[Oct 10 11:08:14] DEBUG[918] chan_sip.c: Stopping retransmission on '
[email protected]' of Request 103: Match Found
[Oct 10 11:08:17] DEBUG[1015] rtp.c: Got RTCP report of 64 bytes[/td]
[td]gibts hier nicht (nur eine Video aus Taste)[/td]
[/tr]
[tr]
[td]auflegen[/td]
[td][Oct 10 11:07:11] DEBUG[1010] channel.c: Didn't get a frame from channel: SIP/6004-00000001
[Oct 10 11:07:11] DEBUG[1010] channel.c: Bridge stops bridging channels SIP/6000-00000000 and SIP/6004-00000001
[Oct 10 11:07:11] DEBUG[1010] channel.c: Hanging up channel 'SIP/6004-00000001'
[Oct 10 11:07:11] DEBUG[1010] chan_sip.c: Hangup call SIP/6004-00000001, SIP callid
[email protected]
[Oct 10 11:07:11] DEBUG[1010] rtp.c: Channel '<unspecified>' has no RTP, not doing anything
[Oct 10 11:07:11] DEBUG[1010] app_dial.c: Exiting with DIALSTATUS=ANSWER.
[Oct 10 11:07:11] DEBUG[1010] pbx.c: Spawn extension (DLPN_DialPlan1,6004,1) exited non-zero on 'SIP/6000-00000000'
[Oct 10 11:07:11] VERBOSE[1010] pbx.c: == Spawn extension (DLPN_DialPlan1, 6004, 1) exited non-zero on 'SIP/6000-00000000'
[Oct 10 11:07:11] DEBUG[1010] channel.c: Soft-Hanging up channel 'SIP/6000-00000000'
[Oct 10 11:07:11] DEBUG[1010] channel.c: Hanging up channel 'SIP/6000-00000000'
[Oct 10 11:07:11] DEBUG[1010] chan_sip.c: Hangup call SIP/6000-00000000, SIP callid
[email protected]
[Oct 10 11:07:11] DEBUG[918] chan_sip.c: Stopping retransmission on '
[email protected]' of Request 102: Match Found[/td]
[td][Oct 10 11:08:22] DEBUG[1015] rtp.c: Got RTCP report of 64 bytes
[Oct 10 11:08:23] DEBUG[1015] channel.c: Didn't get a frame from channel: SIP/6007-00000003
[Oct 10 11:08:23] DEBUG[1015] channel.c: Bridge stops bridging channels SIP/6000-00000002 and SIP/6007-00000003
[Oct 10 11:08:23] DEBUG[1015] channel.c: Hanging up channel 'SIP/6007-00000003'
[Oct 10 11:08:23] DEBUG[1015] chan_sip.c: Hangup call SIP/6007-00000003, SIP callid
[email protected]
[Oct 10 11:08:23] DEBUG[1015] rtp.c: Channel '<unspecified>' has no RTP, not doing anything
[Oct 10 11:08:23] DEBUG[1015] app_dial.c: Exiting with DIALSTATUS=ANSWER.
[Oct 10 11:08:23] DEBUG[1015] pbx.c: Spawn extension (DLPN_DialPlan1,6007,1) exited non-zero on 'SIP/6000-00000002'
[Oct 10 11:08:23] VERBOSE[1015] pbx.c: == Spawn extension (DLPN_DialPlan1, 6007, 1) exited non-zero on 'SIP/6000-00000002'
[Oct 10 11:08:23] DEBUG[1015] channel.c: Soft-Hanging up channel 'SIP/6000-00000002'
[Oct 10 11:08:23] DEBUG[1015] channel.c: Hanging up channel 'SIP/6000-00000002'
[Oct 10 11:08:23] DEBUG[1015] chan_sip.c: Hangup call SIP/6000-00000002, SIP callid
[email protected]
[Oct 10 11:08:23] DEBUG[918] chan_sip.c: Stopping retransmission on '
[email protected]' of Request 102: Match Found[/td]
[td][Oct 10 11:11:47] DEBUG[1024] rtp.c: Got RTCP report of 64 bytes
[Oct 10 11:11:48] DEBUG[1024] channel.c: Didn't get a frame from channel: SIP/6003-00000007
[Oct 10 11:11:48] DEBUG[1024] channel.c: Bridge stops bridging channels SIP/6007-00000006 and SIP/6003-00000007
[Oct 10 11:11:48] DEBUG[1024] channel.c: Hanging up channel 'SIP/6003-00000007'
[Oct 10 11:11:48] DEBUG[1024] chan_sip.c: Hangup call SIP/6003-00000007, SIP callid
[email protected]
[Oct 10 11:11:48] DEBUG[1024] rtp.c: Channel '<unspecified>' has no RTP, not doing anything
[Oct 10 11:11:48] DEBUG[1024] app_dial.c: Exiting with DIALSTATUS=ANSWER.
[Oct 10 11:11:48] DEBUG[1024] pbx.c: Spawn extension (DLPN_DialPlan1,6003,1) exited non-zero on 'SIP/6007-00000006'
[Oct 10 11:11:48] VERBOSE[1024] pbx.c: == Spawn extension (DLPN_DialPlan1, 6003, 1) exited non-zero on 'SIP/6007-00000006'
[Oct 10 11:11:48] DEBUG[1024] channel.c: Soft-Hanging up channel 'SIP/6007-00000006'
[Oct 10 11:11:48] DEBUG[1024] channel.c: Hanging up channel 'SIP/6007-00000006'
[Oct 10 11:11:48] DEBUG[1024] chan_sip.c: Hangup call SIP/6007-00000006, SIP callid
[email protected]
[Oct 10 11:11:48] DEBUG[918] chan_sip.c: Stopping retransmission on '
[email protected]' of Request 102: Match Found
[Oct 10 11:13:38] DEBUG[918] chan_sip.c: Allocating new SIP dialog for
[email protected] - REGISTER (No RTP)
[Oct 10 11:13:38] DEBUG[918] chan_sip.c: Store REGISTER's src-IP
ort for call routing.[/td]
[/tr]
[/table]
