[asterisk-bugs] [JIRA] (ASTERISK-24164) High consume CPU when establishing Conf_bridge
Matt Jordan (JIRA)
noreply at issues.asterisk.org
Wed Aug 6 09:01:56 CDT 2014
[ https://issues.asterisk.org/jira/browse/ASTERISK-24164?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Matt Jordan closed ASTERISK-24164.
----------------------------------
Resolution: Not A Bug
> 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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