[asterisk-bugs] [JIRA] (ASTERISK-24164) High consume CPU when establishing Conf_bridge

Raul Jimeno (JIRA) noreply at issues.asterisk.org
Wed Aug 6 04:21:12 CDT 2014


Raul Jimeno created ASTERISK-24164:
--------------------------------------

             Summary: High consume CPU when establishing Conf_bridge 
                 Key: ASTERISK-24164
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24164
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Bridges/bridge_softmix
    Affects Versions: 11.6.0
         Environment: 2.6.32-431.20.3.el6.x86_64
CentOS release 6.5 (Final)

            Reporter: Raul Jimeno
            Severity: Critical



Hi, 

It is being testing an environment where we have two servers running Asterisk 11.6.0. There is a SIP trunk between both servers. Then, as soon as a "tunnel" between both servers is created the CPU goes up. The more tunnels the more high CPU goes.


Server2_ ->  TNS-ES-SE-TS2 / 172.19.56.24

sip.conf

[toTNS-ES-SE-TS1]
type=peer
host=172.19.56.23
context=InVADEDialler
disallow=all
allow=all
canreinvite=no
insecure=port,invite


Server1_ ->  TNS-ES-SE-TS1 / 172.19.56.23

sip.conf

[toTNS-ES-SE-TS2]
type=peer
host=172.19.56.24
context=InVADEDialler
disallow=all
allow=all
canreinvite=no
insecure=port,invite

confbridge.conf (same in both sites)

[invadeconf_user]
type=user
music_on_hold_class=default
quiet=no
dsp_drop_silence=yes
dsp_talking_threshold=128
dsp_silence_threshold=2000
denoise=yes
jitterbuffer=yes
dtmf_passthrough=yes
announce_join_leave=no

[invadeconf_userq]
type=user
music_on_hold_class=default
quiet=yes
dsp_drop_silence=yes
dsp_talking_threshold=128
dsp_silence_threshold=2000
denoise=yes
jitterbuffer=yes
dtmf_passthrough=yes
announce_join_leave=no

[invadeconf_bridge]
type=bridge
max_members=15
;record_conference=yes
;record_file=</path/to/file>
;internal_sample_rate=auto
;mixing_interval=40
sound_join
sound_leave

extensions.conf  (same in both sites)

[InVADEDialler]
exten => 8000,1,ConfBridge(8000,invadeconf_bridge,invadeconf_userq)
exten => 8001,1,ConfBridge(8001,invadeconf_bridge,invadeconf_userq)
exten => 8002,1,ConfBridge(8002,invadeconf_bridge,invadeconf_userq)


TNS-ES-SE-TS1*CLI> sip show peer toTNS-ES-SE-TS2

  * Name       : toTNS-ES-SE-TS2
  Description  :
  Secret       : <Not set>
  MD5Secret    : <Not set>
  Remote Secret: <Not set>
  Context      : InVADEDialler
  Record On feature : automon
  Record Off feature : automon
  Subscr.Cont. : <Not set>
  Language     :
  Tonezone     : <Not set>
  AMA flags    : Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup    :
  Pickupgroup  :
  Named Callgr :
  Nam. Pickupgr:
  MOH Suggest  :
  Mailbox      :
  VM Extension : asterisk
  LastMsgsSent : 0/0
  Call limit   : 0
  Max forwards : 0
  Dynamic      : No
  Callerid     : "" <>
  MaxCallBR    : 384 kbps
  Expire       : -1
  Insecure     : port,invite
  Force rport  : Auto (No)
  Symmetric RTP: No
  ACL          : No
  DirectMedACL : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: -1
  DirectMedia  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: No
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : No
  Send RPID    : No
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode     : rfc2833
  Timer T1     : 500
  Timer B      : 32000
  ToHost       : 172.19.56.24
  Addr->IP     : 172.19.56.24:5060
  Defaddr->IP  : (null)
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username:
  SIP Options  : (none)
  Codecs       : (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192|silk8|silk12|silk16|silk24)
  Codec Order  : (none)
  Auto-Framing :  No
  Status       : Unmonitored
  Useragent    :
  Reg. Contact :
  Qualify Freq : 60000 ms
  Keepalive    : 0 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess     : 90 secs
  RTP Engine   : asterisk
  Parkinglot   :
  Use Reason   : No
  Encryption   : No


Then, from Asterisk CLI we originate the call to create the tunnel: 

TNS-ES-SE-TS1*CLI> originate SIP/toTNS-ES-SE-TS2/8000 extension 8000 at InVADEDialler

TNS-ES-SE-TS1*CLI> core show channel SIP/toTNS-ES-SE-TS2-00000000
 -- General --
           Name: SIP/toTNS-ES-SE-TS2-00000000
           Type: SIP
       UniqueID: 1407314940.0
       LinkedID: 1407314940.0
      Caller ID: (N/A)
 Caller ID Name: (N/A)
Connected Line ID: (N/A)
Connected Line ID Name: (N/A)
Eff. Connected Line ID: (N/A)
Eff. Connected Line ID Name: (N/A)
    DNID Digits: (N/A)
       Language: en
          State: Up (6)
          Rings: 0
  NativeFormats: (ulaw)
    WriteFormat: slin
     ReadFormat: slin
 WriteTranscode: Yes (slin)->(ulaw)
  ReadTranscode: Yes (ulaw)->(slin)
1st File Descriptor: 150
      Frames in: 106825
     Frames out: 53304
 Time to Hangup: 0
   Elapsed Time: 0h17m46s
  Direct Bridge: <none>
Indirect Bridge: <none>
 --   PBX   --
        Context: InVADEDialler
      Extension: 8000
       Priority: 1
     Call Group: 0
   Pickup Group: 0
    Application: ConfBridge
           Data: 8000,invadeconf_bridge,invadeconf_userq
    Blocking in: ast_waitfor_nandfds
 Call Identifer: [C-00000000]
      Variables:
SIPCALLID=4bae0400094779c253faa5543fecbe5c at 172.19.56.23:5060

  CDR Variables:
level 1: dnid=
level 1: dst=8000
level 1: dcontext=InVADEDialler
level 1: channel=SIP/toTNS-ES-SE-TS2-00000000
level 1: lastapp=ConfBridge
level 1: lastdata=8000,invadeconf_bridge,invadeconf_userq
level 1: start=2014-08-06 10:49:00
level 1: answer=2014-08-06 10:49:00
level 1: duration=1066
level 1: billsec=1066
level 1: disposition=ANSWERED
level 1: amaflags=DOCUMENTATION
level 1: uniqueid=1407314940.0
level 1: linkedid=1407314940.0
level 1: sequence=0

Asterisk_Logs

[Aug  6 11:13:25] Asterisk 11.6.0 built by root @ TNS-ES-SE-TS2 on a x86_64 running Linux on 2014-08-05 11:50:27 UTC
[Aug  6 11:13:25] DEBUG[2415] config.c: Parsing /etc/asterisk/logger.conf
[Aug  6 11:13:25] VERBOSE[2415] config.c:   == Parsing '/etc/asterisk/logger.conf': Found
[Aug  6 11:13:25] VERBOSE[2415] logger.c:  Asterisk Queue Logger restarted
[Aug  6 11:13:26] VERBOSE[1932] chan_sip.c:
ÿ<--- SIP read from UDP:172.19.58.152:5061 --->
ÿ
ÿ
ÿ<------------->
[Aug  6 11:13:26] DEBUG[1932] chan_sip.c:  Header  0 [  0]:
[Aug  6 11:13:37] VERBOSE[1932] chan_sip.c:
ÿ<--- SIP read from UDP:172.19.56.23:5060 --->
ÿINVITE sip:8000 at 172.19.56.24 SIP/2.0
ÿVia: SIP/2.0/UDP 172.19.56.23:5060;branch=z9hG4bK36fd679a
ÿMax-Forwards: 70
ÿFrom: "Anonymous" <sip:anonymous at anonymous.invalid>;tag=as6448840f
ÿTo: <sip:8000 at 172.19.56.24>
ÿContact: <sip:anonymous at 172.19.56.23:5060>
ÿCall-ID: 4ea491311decb52c17b4ac1d28d9f46e at 172.19.56.23:5060
ÿCSeq: 102 INVITE
ÿUser-Agent: Asterisk PBX 11.6.0
ÿDate: Wed, 06 Aug 2014 09:13:37 GMT
ÿAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
ÿSupported: replaces, timer
ÿContent-Type: application/sdp
ÿContent-Length: 1309
ÿ
ÿv=0
ÿo=root 647713105 647713105 IN IP4 172.19.56.23
ÿs=Asterisk PBX 11.6.0
ÿc=IN IP4 172.19.56.23
ÿt=0 0
ÿm=audio 13108 RTP/AVP 10 4 3 0 8 112 5 7 18 110 97 111 9 102 115 116 117 96 100 107 108 118 119 101
ÿa=rtpmap:10 L16/8000
ÿa=rtpmap:4 G723/8000
ÿa=fmtp:4 annexa=no
ÿa=rtpmap:3 GSM/8000
ÿa=rtpmap:0 PCMU/8000
ÿa=rtpmap:8 PCMA/8000
ÿa=rtpmap:112 AAL2-G726-32/8000
ÿa=rtpmap:5 DVI4/8000
ÿa=rtpmap:7 LPC/8000
ÿa=rtpmap:18 G729/8000
ÿa=fmtp:18 annexb=no
ÿa=rtpmap:110 speex/8000
ÿa=rtpmap:97 iLBC/8000
ÿa=fmtp:97 mode=30
ÿa=rtpmap:111 G726-32/8000
ÿa=rtpmap:9 G722/8000
ÿa=rtpmap:102 G7221/16000
ÿa=fmtp:102 bitrate=32000
ÿa=rtpmap:115 G7221/32000
ÿa=fmtp:115 bitrate=48000
ÿa=rtpmap:116 G719/48000
ÿa=fmtp:116 bitrate=64000
ÿa=rtpmap:117 speex/16000
ÿa=rtpmap:96 SILK/8000
ÿa=fmtp:96 maxaveragebitrate=10000
ÿa=fmtp:96 usedtx=0
ÿa=fmtp:96 useinbandfec=1
ÿa=rtpmap:100 SILK/12000
ÿa=fmtp:100 maxaveragebitrate=12000
ÿa=fmtp:100 usedtx=0
ÿa=fmtp:100 useinbandfec=1
ÿa=rtpmap:107 SILK/16000
ÿa=fmtp:107 maxaveragebitrate=20000
ÿa=fmtp:107 usedtx=0
ÿa=fmtp:107 useinbandfec=1
ÿa=rtpmap:108 SILK/24000
ÿa=fmtp:108 maxaveragebitrate=30000
ÿa=fmtp:108 usedtx=0
ÿa=fmtp:108 useinbandfec=1
ÿa=rtpmap:118 L16/16000
ÿa=rtpmap:119 speex/32000
ÿa=rtpmap:101 telephone-event/8000
ÿa=fmtp:101 0-16
ÿa=ptime:20
ÿa=sendrecv
ÿ<------------->
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header  0 [ 36]: INVITE sip:8000 at 172.19.56.24 SIP/2.0
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header  1 [ 57]: Via: SIP/2.0/UDP 172.19.56.23:5060;branch=z9hG4bK36fd679a
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header  2 [ 16]: Max-Forwards: 70
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header  3 [ 66]: From: "Anonymous" <sip:anonymous at anonymous.invalid>;tag=as6448840f
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header  4 [ 27]: To: <sip:8000 at 172.19.56.24>
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header  5 [ 42]: Contact: <sip:anonymous at 172.19.56.23:5060>
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header  6 [ 59]: Call-ID: 4ea491311decb52c17b4ac1d28d9f46e at 172.19.56.23:5060
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header  7 [ 16]: CSeq: 102 INVITE
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header  8 [ 31]: User-Agent: Asterisk PBX 11.6.0
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header  9 [ 35]: Date: Wed, 06 Aug 2014 09:13:37 GMT
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header 10 [ 81]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header 11 [ 26]: Supported: replaces, timer
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header 12 [ 29]: Content-Type: application/sdp
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header 13 [ 20]: Content-Length: 1309
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header 14 [  0]:
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body  0 [  3]: v=0
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body  1 [ 46]: o=root 647713105 647713105 IN IP4 172.19.56.23
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body  2 [ 21]: s=Asterisk PBX 11.6.0
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body  3 [ 21]: c=IN IP4 172.19.56.23
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body  4 [  5]: t=0 0
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body  5 [ 99]: m=audio 13108 RTP/AVP 10 4 3 0 8 112 5 7 18 110 97 111 9 102 115 116 117 96 100 107 108 118 119 101
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body  6 [ 20]: a=rtpmap:10 L16/8000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body  7 [ 20]: a=rtpmap:4 G723/8000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body  8 [ 18]: a=fmtp:4 annexa=no
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body  9 [ 19]: a=rtpmap:3 GSM/8000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 10 [ 20]: a=rtpmap:0 PCMU/8000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 11 [ 20]: a=rtpmap:8 PCMA/8000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 12 [ 30]: a=rtpmap:112 AAL2-G726-32/8000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 13 [ 20]: a=rtpmap:5 DVI4/8000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 14 [ 19]: a=rtpmap:7 LPC/8000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 15 [ 21]: a=rtpmap:18 G729/8000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 16 [ 19]: a=fmtp:18 annexb=no
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 17 [ 23]: a=rtpmap:110 speex/8000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 18 [ 21]: a=rtpmap:97 iLBC/8000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 19 [ 17]: a=fmtp:97 mode=30
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 20 [ 25]: a=rtpmap:111 G726-32/8000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 21 [ 20]: a=rtpmap:9 G722/8000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 22 [ 24]: a=rtpmap:102 G7221/16000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 23 [ 24]: a=fmtp:102 bitrate=32000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 24 [ 24]: a=rtpmap:115 G7221/32000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 25 [ 24]: a=fmtp:115 bitrate=48000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 26 [ 23]: a=rtpmap:116 G719/48000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 27 [ 24]: a=fmtp:116 bitrate=64000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 28 [ 24]: a=rtpmap:117 speex/16000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 29 [ 21]: a=rtpmap:96 SILK/8000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 30 [ 33]: a=fmtp:96 maxaveragebitrate=10000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 31 [ 18]: a=fmtp:96 usedtx=0
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 32 [ 24]: a=fmtp:96 useinbandfec=1
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 33 [ 23]: a=rtpmap:100 SILK/12000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 34 [ 34]: a=fmtp:100 maxaveragebitrate=12000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 35 [ 19]: a=fmtp:100 usedtx=0
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 36 [ 25]: a=fmtp:100 useinbandfec=1
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 37 [ 23]: a=rtpmap:107 SILK/16000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 38 [ 34]: a=fmtp:107 maxaveragebitrate=20000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 39 [ 19]: a=fmtp:107 usedtx=0
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 40 [ 25]: a=fmtp:107 useinbandfec=1
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 41 [ 23]: a=rtpmap:108 SILK/24000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 42 [ 34]: a=fmtp:108 maxaveragebitrate=30000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 43 [ 19]: a=fmtp:108 usedtx=0
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 44 [ 25]: a=fmtp:108 useinbandfec=1
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 45 [ 22]: a=rtpmap:118 L16/16000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 46 [ 24]: a=rtpmap:119 speex/32000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 47 [ 33]: a=rtpmap:101 telephone-event/8000
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 48 [ 15]: a=fmtp:101 0-16
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 49 [ 10]: a=ptime:20
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:    Body 50 [ 10]: a=sendrecv
[Aug  6 11:13:37] VERBOSE[1932] chan_sip.c: --- (14 headers 51 lines) ---
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c: = Looking for  Call ID: 4ea491311decb52c17b4ac1d28d9f46e at 172.19.56.23:5060 (Checking From) --From tag as6448840f --To-tag
[Aug  6 11:13:37] DEBUG[1932] acl.c: For destination '172.19.56.23', our source address is '172.19.56.24'.
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 172.19.56.24:5060
[Aug  6 11:13:37] DEBUG[1932] netsock2.c: Splitting '172.19.56.23:5060' into...
[Aug  6 11:13:37] DEBUG[1932] netsock2.c: ...host '172.19.56.23' and port '5060'.
[Aug  6 11:13:37] VERBOSE[1932] chan_sip.c: Sending to 172.19.56.23:5060 (no NAT)
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c: Allocating new SIP dialog for 4ea491311decb52c17b4ac1d28d9f46e at 172.19.56.23:5060 - INVITE (No RTP)
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Aug  6 11:13:37] DEBUG[1932][C-00000002] sip/reqresp_parser.c: Begin: parsing SIP "Supported: replaces, timer"
[Aug  6 11:13:37] DEBUG[1932][C-00000002] sip/reqresp_parser.c: Found SIP option: -replaces-
[Aug  6 11:13:37] DEBUG[1932][C-00000002] sip/reqresp_parser.c: Matched SIP option: replaces
[Aug  6 11:13:37] DEBUG[1932][C-00000002] sip/reqresp_parser.c: Found SIP option: -timer-
[Aug  6 11:13:37] DEBUG[1932][C-00000002] sip/reqresp_parser.c: Matched SIP option: timer
[Aug  6 11:13:37] DEBUG[1932][C-00000002] netsock2.c: Splitting '172.19.56.23:5060' into...
[Aug  6 11:13:37] DEBUG[1932][C-00000002] netsock2.c: ...host '172.19.56.23' and port '5060'.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Sending to 172.19.56.23:5060 (no NAT)
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Initializing initreq for method INVITE - callid 4ea491311decb52c17b4ac1d28d9f46e at 172.19.56.23:5060
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Using INVITE request as basis request - 4ea491311decb52c17b4ac1d28d9f46e at 172.19.56.23:5060
[Aug  6 11:13:37] DEBUG[1932][C-00000002] netsock2.c: Splitting 'anonymous.invalid' into...
[Aug  6 11:13:37] DEBUG[1932][C-00000002] netsock2.c: ...host 'anonymous.invalid' and port ''.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found peer 'toTNS-ES-SE-TS1' for 'anonymous' from 172.19.56.23:5060
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x7fc54c057128'
[Aug  6 11:13:37] DEBUG[1932][C-00000002] res_rtp_asterisk.c: Allocated port 11072 for RTP instance '0x7fc54c057128'
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: RTP instance '0x7fc54c057128' is setup and ready to go
[Aug  6 11:13:37] DEBUG[1932][C-00000002] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x7fc54c057128'
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] netsock2.c:   == Using SIP RTP CoS mark 5
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Setting NAT on RTP to Off
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED OR FAILED.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing session-level SDP o=root 647713105 647713105 IN IP4 172.19.56.23... OK.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing session-level SDP s=Asterisk PBX 11.6.0... UNSUPPORTED OR FAILED.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] netsock2.c: Splitting '172.19.56.23' into...
[Aug  6 11:13:37] DEBUG[1932][C-00000002] netsock2.c: ...host '172.19.56.23' and port ''.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing session-level SDP c=IN IP4 172.19.56.23... OK.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found RTP audio format 10
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Setting payload 10 based on m type on 0x7fc55b7d7bc0
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found RTP audio format 4
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Setting payload 4 based on m type on 0x7fc55b7d7bc0
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found RTP audio format 3
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Setting payload 3 based on m type on 0x7fc55b7d7bc0
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found RTP audio format 0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Setting payload 0 based on m type on 0x7fc55b7d7bc0
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found RTP audio format 8
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Setting payload 8 based on m type on 0x7fc55b7d7bc0
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found RTP audio format 112
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Setting payload 112 based on m type on 0x7fc55b7d7bc0
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found RTP audio format 5
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Setting payload 5 based on m type on 0x7fc55b7d7bc0
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found RTP audio format 7
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Setting payload 7 based on m type on 0x7fc55b7d7bc0
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found RTP audio format 18
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Setting payload 18 based on m type on 0x7fc55b7d7bc0
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found RTP audio format 110
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Setting payload 110 based on m type on 0x7fc55b7d7bc0
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found RTP audio format 97
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Setting payload 97 based on m type on 0x7fc55b7d7bc0
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found RTP audio format 111
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Setting payload 111 based on m type on 0x7fc55b7d7bc0
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found RTP audio format 9
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Setting payload 9 based on m type on 0x7fc55b7d7bc0
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found RTP audio format 102
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Setting payload 102 based on m type on 0x7fc55b7d7bc0
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found RTP audio format 115
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Setting payload 115 based on m type on 0x7fc55b7d7bc0
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found RTP audio format 116
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Setting payload 116 based on m type on 0x7fc55b7d7bc0
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found RTP audio format 117
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Setting payload 117 based on m type on 0x7fc55b7d7bc0
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found RTP audio format 96
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Setting payload 96 based on m type on 0x7fc55b7d7bc0
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found RTP audio format 100
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Setting payload 100 based on m type on 0x7fc55b7d7bc0
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found RTP audio format 107
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Setting payload 107 based on m type on 0x7fc55b7d7bc0
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found RTP audio format 108
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Setting payload 108 based on m type on 0x7fc55b7d7bc0
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found RTP audio format 118
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Setting payload 118 based on m type on 0x7fc55b7d7bc0
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found RTP audio format 119
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Setting payload 119 based on m type on 0x7fc55b7d7bc0
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found RTP audio format 101
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Setting payload 101 based on m type on 0x7fc55b7d7bc0
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found audio description format L16 for ID 10
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:10 L16/8000... OK.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found audio description format G723 for ID 4
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:4 G723/8000... OK.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:4 annexa=no... OK.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found audio description format GSM for ID 3
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:3 GSM/8000... OK.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found audio description format PCMU for ID 0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found audio description format PCMA for ID 8
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found audio description format AAL2-G726-32 for ID 112
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:112 AAL2-G726-32/8000... OK.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found audio description format DVI4 for ID 5
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:5 DVI4/8000... OK.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found audio description format LPC for ID 7
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:7 LPC/8000... OK.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found audio description format G729 for ID 18
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729/8000... OK.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:18 annexb=no... OK.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found audio description format speex for ID 110
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:110 speex/8000... OK.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found audio description format iLBC for ID 97
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 iLBC/8000... OK.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:97 mode=30... OK.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found audio description format G726-32 for ID 111
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:111 G726-32/8000... OK.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found audio description format G722 for ID 9
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:9 G722/8000... OK.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found audio description format G7221 for ID 102
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:102 G7221/16000... OK.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:102 bitrate=32000... OK.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found audio description format G7221 for ID 115
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:115 G7221/32000... OK.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:115 bitrate=48000... OK.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found audio description format G719 for ID 116
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:116 G719/48000... OK.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:116 bitrate=64000... OK.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found audio description format speex for ID 117
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:117 speex/16000... OK.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found audio description format SILK for ID 96
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 SILK/8000... OK.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 maxaveragebitrate=10000... OK.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 usedtx=0... OK.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:96 useinbandfec=1... OK.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found audio description format SILK for ID 100
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:100 SILK/12000... OK.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:100 maxaveragebitrate=12000... OK.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:100 usedtx=0... OK.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:100 useinbandfec=1... OK.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found audio description format SILK for ID 107
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:107 SILK/16000... OK.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:107 maxaveragebitrate=20000... OK.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:107 usedtx=0... OK.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:107 useinbandfec=1... OK.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found audio description format SILK for ID 108
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:108 SILK/24000... OK.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:108 maxaveragebitrate=30000... OK.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:108 usedtx=0... OK.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:108 useinbandfec=1... OK.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found audio description format L16 for ID 118
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:118 L16/16000... OK.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found audio description format speex for ID 119
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:119 speex/32000... OK.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Found audio description format telephone-event for ID 101
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-16... UNSUPPORTED OR FAILED.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=ptime:20... OK.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Capabilities: us - (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192|silk8|silk12|silk16|silk24), peer - audio=(g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|siren7|siren14|g719|speex32|silk8|silk12|silk16|silk24)/video=(nothing)/text=(nothing), combined - (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|siren7|siren14|g719|speex32|silk8|silk12|silk16|silk24)
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Aug  6 11:13:37] DEBUG[1932][C-00000002] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7fc54c057128'
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Peer audio RTP is at port 172.19.56.23:13108
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Copying payload 0 from 0x7fc55b7d7bc0 to 0x7fc54c0572f0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Copying payload 3 from 0x7fc55b7d7bc0 to 0x7fc54c0572f0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Copying payload 4 from 0x7fc55b7d7bc0 to 0x7fc54c0572f0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Copying payload 5 from 0x7fc55b7d7bc0 to 0x7fc54c0572f0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Copying payload 7 from 0x7fc55b7d7bc0 to 0x7fc54c0572f0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Copying payload 8 from 0x7fc55b7d7bc0 to 0x7fc54c0572f0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Copying payload 9 from 0x7fc55b7d7bc0 to 0x7fc54c0572f0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Copying payload 10 from 0x7fc55b7d7bc0 to 0x7fc54c0572f0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Copying payload 18 from 0x7fc55b7d7bc0 to 0x7fc54c0572f0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Copying payload 96 from 0x7fc55b7d7bc0 to 0x7fc54c0572f0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Copying payload 97 from 0x7fc55b7d7bc0 to 0x7fc54c0572f0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Copying payload 100 from 0x7fc55b7d7bc0 to 0x7fc54c0572f0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Copying payload 101 from 0x7fc55b7d7bc0 to 0x7fc54c0572f0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Copying payload 102 from 0x7fc55b7d7bc0 to 0x7fc54c0572f0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Copying payload 107 from 0x7fc55b7d7bc0 to 0x7fc54c0572f0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Copying payload 108 from 0x7fc55b7d7bc0 to 0x7fc54c0572f0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Copying payload 110 from 0x7fc55b7d7bc0 to 0x7fc54c0572f0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Copying payload 111 from 0x7fc55b7d7bc0 to 0x7fc54c0572f0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Copying payload 112 from 0x7fc55b7d7bc0 to 0x7fc54c0572f0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Copying payload 115 from 0x7fc55b7d7bc0 to 0x7fc54c0572f0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Copying payload 116 from 0x7fc55b7d7bc0 to 0x7fc54c0572f0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Copying payload 117 from 0x7fc55b7d7bc0 to 0x7fc54c0572f0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Copying payload 118 from 0x7fc55b7d7bc0 to 0x7fc54c0572f0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] rtp_engine.c: Copying payload 119 from 0x7fc55b7d7bc0 to 0x7fc54c0572f0
[Aug  6 11:13:37] DEBUG[1932][C-00000002] res_rtp_asterisk.c: Ignoring duplicate RTCP property on RTP instance '0x7fc54c057128'
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: We're settling with these formats: (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|siren7|siren14|g719|speex32|silk8|silk12|silk16|silk24)
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Checking SIP call limits for device
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Updating call counter for incoming call
[Aug  6 11:13:37] DEBUG[1932][C-00000002] netsock2.c: Splitting '172.19.56.24' into...
[Aug  6 11:13:37] DEBUG[1932][C-00000002] netsock2.c: ...host '172.19.56.24' and port ''.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] netsock2.c: Splitting 'anonymous.invalid' into...
[Aug  6 11:13:37] DEBUG[1932][C-00000002] netsock2.c: ...host 'anonymous.invalid' and port ''.
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: Looking for 8000 in InVADEDialler (domain 172.19.56.24)
[Aug  6 11:13:37] DEBUG[1932][C-00000002] format_pref.c: Could not find preferred codec - Going for the best codec
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: *** Our native formats are (ulaw)
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: *** Joint capabilities are (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|siren7|siren14|g719|speex32|silk8|silk12|silk16|silk24)
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: *** Our capabilities are (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192|silk8|silk12|silk16|silk24)
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: *** AST_CODEC_CHOOSE formats are ulaw
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: This channel will not be able to handle video.
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: build_route: Contact hop: <sip:anonymous at 172.19.56.23:5060>
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c: list_route: hop: <sip:anonymous at 172.19.56.23:5060>
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Incoming INVITE with 'timer' option supported
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Session timer started: 129 - 4ea491311decb52c17b4ac1d28d9f46e at 172.19.56.23:5060 900000ms
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: SIP/toTNS-ES-SE-TS1-00000002: New call is still down.... Trying...
[Aug  6 11:13:37] VERBOSE[1932][C-00000002] chan_sip.c:
ÿ<--- Transmitting (no NAT) to 172.19.56.23:5060 --->
ÿSIP/2.0 100 Trying
ÿVia: SIP/2.0/UDP 172.19.56.23:5060;branch=z9hG4bK36fd679a;received=172.19.56.23
ÿFrom: "Anonymous" <sip:anonymous at anonymous.invalid>;tag=as6448840f
ÿTo: <sip:8000 at 172.19.56.24>
ÿCall-ID: 4ea491311decb52c17b4ac1d28d9f46e at 172.19.56.23:5060
ÿCSeq: 102 INVITE
ÿServer: Asterisk PBX 11.6.0
ÿAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
ÿSupported: replaces, timer
ÿSession-Expires: 1800;refresher=uas
ÿContact: <sip:8000 at 172.19.56.24:5060>
ÿContent-Length: 0
ÿ
ÿ
ÿ<------------>
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 172.19.56.23:5060
[Aug  6 11:13:37] DEBUG[1924] devicestate.c: No provider found, checking channel drivers for SIP - toTNS-ES-SE-TS1
[Aug  6 11:13:37] DEBUG[2418][C-00000002] pbx.c: Launching 'ConfBridge'
[Aug  6 11:13:37] DEBUG[1924] chan_sip.c: Checking device state for peer toTNS-ES-SE-TS1
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] pbx.c:     -- Executing [8000 at InVADEDialler:1] ConfBridge("SIP/toTNS-ES-SE-TS1-00000002", "8000,invadeconf_bridge,invadeconf_userq") in new stack
[Aug  6 11:13:37] DEBUG[1924] devicestate.c: Changing state for SIP/toTNS-ES-SE-TS1 - state 1 (Not in use)
[Aug  6 11:13:37] DEBUG[1924] devicestate.c: device 'SIP/toTNS-ES-SE-TS1' state '1'
[Aug  6 11:13:37] DEBUG[1958] app_queue.c: Device 'SIP/toTNS-ES-SE-TS1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Aug  6 11:13:37] DEBUG[1924] devicestate.c: No provider found, checking channel drivers for SIP - toTNS-ES-SE-TS1
[Aug  6 11:13:37] DEBUG[2418][C-00000002] chan_sip.c: SIP answering channel: SIP/toTNS-ES-SE-TS1-00000002
[Aug  6 11:13:37] DEBUG[1924] chan_sip.c: Checking device state for peer toTNS-ES-SE-TS1
[Aug  6 11:13:37] DEBUG[2418][C-00000002] res_rtp_asterisk.c: Setting the marker bit due to a source update
[Aug  6 11:13:37] DEBUG[1924] devicestate.c: Changing state for SIP/toTNS-ES-SE-TS1 - state 1 (Not in use)
[Aug  6 11:13:37] DEBUG[2418][C-00000002] chan_sip.c: Setting framing from config on incoming call
[Aug  6 11:13:37] DEBUG[1924] devicestate.c: device 'SIP/toTNS-ES-SE-TS1' state '1'
[Aug  6 11:13:37] DEBUG[1958] app_queue.c: Device 'SIP/toTNS-ES-SE-TS1' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
[Aug  6 11:13:37] DEBUG[2418][C-00000002] chan_sip.c: ** Our capability: (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|siren7|siren14|g719|speex32|silk8|silk12|silk16|silk24) Video flag: True Text flag: True
[Aug  6 11:13:37] DEBUG[2418][C-00000002] chan_sip.c: ** Our prefcodec: (nothing)
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Audio is at 11072
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Adding codec 100001 (g723) to SDP
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Adding codec 100002 (gsm) to SDP
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Adding codec 100004 (alaw) to SDP
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Adding codec 100005 (g726aal2) to SDP
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Adding codec 100006 (adpcm) to SDP
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Adding codec 100007 (lpc10) to SDP
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Adding codec 100008 (g729) to SDP
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Adding codec 100009 (speex) to SDP
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Adding codec 100010 (ilbc) to SDP
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Adding codec 100011 (g726) to SDP
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Adding codec 100012 (g722) to SDP
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Adding codec 100013 (siren7) to SDP
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Adding codec 100014 (siren14) to SDP
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Adding codec 100015 (g719) to SDP
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Adding codec 100016 (speex16) to SDP
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Adding codec 100018 (silk8) to SDP
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Adding codec 100018 (silk12) to SDP
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Adding codec 100018 (silk16) to SDP
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Adding codec 100018 (silk24) to SDP
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Adding codec 100019 (slin) to SDP
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Adding codec 100021 (slin16) to SDP
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Adding codec 100028 (speex32) to SDP
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Aug  6 11:13:37] DEBUG[2418][C-00000002] chan_sip.c: -- Done with adding codecs to SDP
[Aug  6 11:13:37] DEBUG[2418][C-00000002] chan_sip.c: Done building SDP. Settling with this capability: (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|siren7|siren14|g719|speex32|silk8|silk12|silk16|silk24)
[Aug  6 11:13:37] VERBOSE[2418][C-00000002] chan_sip.c:
ÿ<--- Reliably Transmitting (no NAT) to 172.19.56.23:5060 --->
ÿSIP/2.0 200 OK
ÿVia: SIP/2.0/UDP 172.19.56.23:5060;branch=z9hG4bK36fd679a;received=172.19.56.23
ÿFrom: "Anonymous" <sip:anonymous at anonymous.invalid>;tag=as6448840f
ÿTo: <sip:8000 at 172.19.56.24>;tag=as65e068c9
ÿCall-ID: 4ea491311decb52c17b4ac1d28d9f46e at 172.19.56.23:5060
ÿCSeq: 102 INVITE
ÿServer: Asterisk PBX 11.6.0
ÿAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
ÿSupported: replaces, timer
ÿSession-Expires: 1800;refresher=uas
ÿContact: <sip:8000 at 172.19.56.24:5060>
ÿContent-Type: application/sdp
ÿRequire: timer
ÿContent-Length: 1309
ÿ
ÿv=0
ÿo=root 946938154 946938154 IN IP4 172.19.56.24
ÿs=Asterisk PBX 11.6.0
ÿc=IN IP4 172.19.56.24
ÿt=0 0
ÿm=audio 11072 RTP/AVP 4 3 0 8 112 5 7 18 110 97 111 9 102 115 116 117 96 100 107 108 10 118 119 101
ÿa=rtpmap:4 G723/8000
ÿa=fmtp:4 annexa=no
ÿa=rtpmap:3 GSM/8000
ÿa=rtpmap:0 PCMU/8000
ÿa=rtpmap:8 PCMA/8000
ÿa=rtpmap:112 AAL2-G726-32/8000
ÿa=rtpmap:5 DVI4/8000
ÿa=rtpmap:7 LPC/8000
ÿa=rtpmap:18 G729/8000
ÿa=fmtp:18 annexb=no
ÿa=rtpmap:110 speex/8000
ÿa=rtpmap:97 iLBC/8000
ÿa=fmtp:97 mode=30
ÿa=rtpmap:111 G726-32/8000
ÿa=rtpmap:9 G722/8000
ÿa=rtpmap:102 G7221/16000
ÿa=fmtp:102 bitrate=32000
ÿa=rtpmap:115 G7221/32000
ÿa=fmtp:115 bitrate=48000
ÿa=rtpmap:116 G719/48000
ÿa=fmtp:116 bitrate=64000
ÿa=rtpmap:117 speex/16000
ÿa=rtpmap:96 SILK/8000
ÿa=fmtp:96 maxaveragebitrate=10000
ÿa=fmtp:96 usedtx=0
ÿa=fmtp:96 useinbandfec=1
ÿa=rtpmap:100 SILK/12000
ÿa=fmtp:100 maxaveragebitrate=12000
ÿa=fmtp:100 usedtx=0
ÿa=fmtp:100 useinbandfec=1
ÿa=rtpmap:107 SILK/16000
ÿa=fmtp:107 maxaveragebitrate=20000
ÿa=fmtp:107 usedtx=0
ÿa=fmtp:107 useinbandfec=1
ÿa=rtpmap:108 SILK/24000
ÿa=fmtp:108 maxaveragebitrate=30000
ÿa=fmtp:108 usedtx=0
ÿa=fmtp:108 useinbandfec=1
ÿa=rtpmap:10 L16/8000
ÿa=rtpmap:118 L16/16000
ÿa=rtpmap:119 speex/32000
ÿa=rtpmap:101 telephone-event/8000
ÿa=fmtp:101 0-16
ÿa=ptime:20
ÿa=sendrecv
ÿ
ÿ<------------>
[Aug  6 11:13:37] DEBUG[2418][C-00000002] chan_sip.c: *** SIP TIMER: Initializing retransmit timer on packet: Id  #131
[Aug  6 11:13:37] DEBUG[2418][C-00000002] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 172.19.56.23:5060
[Aug  6 11:13:37] VERBOSE[1932] chan_sip.c:
ÿ<--- SIP read from UDP:172.19.56.23:5060 --->
ÿACK sip:8000 at 172.19.56.24:5060 SIP/2.0
ÿVia: SIP/2.0/UDP 172.19.56.23:5060;branch=z9hG4bK75ad3f09
ÿMax-Forwards: 70
ÿFrom: "Anonymous" <sip:anonymous at anonymous.invalid>;tag=as6448840f
ÿTo: <sip:8000 at 172.19.56.24>;tag=as65e068c9
ÿContact: <sip:anonymous at 172.19.56.23:5060>
ÿCall-ID: 4ea491311decb52c17b4ac1d28d9f46e at 172.19.56.23:5060
ÿCSeq: 102 ACK
ÿUser-Agent: Asterisk PBX 11.6.0
ÿContent-Length: 0
ÿ
ÿ<------------->
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header  0 [ 38]: ACK sip:8000 at 172.19.56.24:5060 SIP/2.0
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header  1 [ 57]: Via: SIP/2.0/UDP 172.19.56.23:5060;branch=z9hG4bK75ad3f09
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header  2 [ 16]: Max-Forwards: 70
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header  3 [ 66]: From: "Anonymous" <sip:anonymous at anonymous.invalid>;tag=as6448840f
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header  4 [ 42]: To: <sip:8000 at 172.19.56.24>;tag=as65e068c9
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header  5 [ 42]: Contact: <sip:anonymous at 172.19.56.23:5060>
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header  6 [ 59]: Call-ID: 4ea491311decb52c17b4ac1d28d9f46e at 172.19.56.23:5060
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header  7 [ 13]: CSeq: 102 ACK
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header  8 [ 31]: User-Agent: Asterisk PBX 11.6.0
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c:  Header  9 [ 17]: Content-Length: 0
[Aug  6 11:13:37] VERBOSE[1932] chan_sip.c: --- (10 headers 0 lines) ---
[Aug  6 11:13:37] DEBUG[1932] chan_sip.c: = Looking for  Call ID: 4ea491311decb52c17b4ac1d28d9f46e at 172.19.56.23:5060 (Checking From) --From tag as6448840f --To-tag as65e068c9
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #131
[Aug  6 11:13:37] DEBUG[1932][C-00000002] chan_sip.c: Stopping retransmission on '4ea491311decb52c17b4ac1d28d9f46e at 172.19.56.23:5060' of Response 102: Match Found
[Aug  6 11:13:37] DEBUG[2418][C-00000002] res_rtp_asterisk.c: 0x7fc54c016820 -- Probation learning mode pass with source address 172.19.56.23:13108
[Aug  6 11:13:37] DEBUG[2418][C-00000002] app_confbridge.c: Trying to find conference bridge '8000'
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Bridge technology simple_bridge does not have the capabilities we need.
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Bridge technology multiplexed_bridge does not have the capabilities we need.
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Chose bridge technology softmix
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Giving bridge technology softmix the bridge structure 0x7fc4cc001cb8 to setup
[Aug  6 11:13:37] DEBUG[2418][C-00000002] app_confbridge.c: Created conference '8000' and linked to container.
[Aug  6 11:13:37] DEBUG[2418][C-00000002] devicestate.c: device 'confbridge:8000' state '2'
[Aug  6 11:13:37] DEBUG[2418][C-00000002] confbridge/conf_state.c: Changing conference '8000' state from EMPTY to SINGLE
[Aug  6 11:13:37] DEBUG[1958] app_queue.c: Device 'confbridge:8000' changed to state '2' (In use) but we don't care because they're not a member of any queue.
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Joining bridge channel 0x7fc4cc004528 to bridge 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Added channel SIP/toTNS-ES-SE-TS1-00000002(0x7fc54c019168) to bridge array on 0x7fc4cc001cb8, new count is 1
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Bridge technology softmix wants to read any of formats (slin) but channel has ulaw
[Aug  6 11:13:37] DEBUG[2418][C-00000002] channel.c: Set channel SIP/toTNS-ES-SE-TS1-00000002 to read format slin
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Bridge 0x7fc4cc001cb8 put channel SIP/toTNS-ES-SE-TS1-00000002 into read format slin
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Bridge technology softmix wants to write any of formats (slin) but channel has ulaw
[Aug  6 11:13:37] DEBUG[2418][C-00000002] channel.c: Set channel SIP/toTNS-ES-SE-TS1-00000002 to write format slin
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Bridge 0x7fc4cc001cb8 put channel SIP/toTNS-ES-SE-TS1-00000002 into write format slin
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Giving bridge technology softmix notification that 0x7fc4cc004528 is joining bridge 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2418][C-00000002] dsp.c: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21
[Aug  6 11:13:37] DEBUG[2418][C-00000002] dsp.c: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Starting a bridge thread for bridge 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2419][C-00000002] bridging.c: Started bridge thread for 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2419][C-00000002] bridging.c: Launching bridge thread function 0x7fc55fc1d5d0 for bridge 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2418][C-00000002] res_rtp_asterisk.c: Ooh, format changed from unknown to ulaw
[Aug  6 11:13:37] DEBUG[2418][C-00000002] res_rtp_asterisk.c: Created smoother: format: ulaw ms: 20 len: 160
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:37] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:38] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:38] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:38] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:38] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8
[Aug  6 11:13:38] DEBUG[2418][C-00000002] bridging.c: Going into a multithreaded waitfor for bridge channel 0x7fc4cc004528 of bridge 0x7fc4cc001cb8






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