Шлется видео, если есть запрет видео у хоста

[general]
context=4tozanah
disallow=all
allow=alaw
allow=ulaw
allow=all
bindpaddr=0.0.0.0
bindport=5060
dtmfmode=rfc2833
allowguest=no
canreinvite=no
maxexpiry=3600
checkmwi=0

[4321]
secret=****
callerid=1234
type=friend
qualify=yes
nat=yes
host=dynamic
dtmfmode=rfc2833
context=users
canreinvite=no
videosupport=no

[4321]
secret=*
callerid=4321
type=friend
qualify=yes
nat=yes
host=dynamic
dtmfmode=rfc2833
context=users
canreinvite=no
videosupport=no

[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: = Looking for Call ID: ec87bd10-24943216@192.168.217.179 (Checking From) --From tag ee6fd0c82b6871beo0 --To-tag
[Dec 22 08:37:36] DEBUG[13050] acl.c: For destination '192.168.217.179', our source address is '192.168.217.11'.
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.217.11:5060
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Allocating new SIP dialog for ec87bd10-24943216@192.168.217.179 - INVITE (No RTP)
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Dec 22 08:37:36] DEBUG[13050] sip/reqresp_parser.c: Begin: parsing SIP "Supported: x-sipura"
[Dec 22 08:37:36] DEBUG[13050] sip/reqresp_parser.c: Found SIP option: -x-sipura-
[Dec 22 08:37:36] DEBUG[13050] sip/reqresp_parser.c: Found private SIP option, not supported: x-sipura
[Dec 22 08:37:36] DEBUG[13050] netsock2.c: Splitting '192.168.217.179:5060' gives...
[Dec 22 08:37:36] DEBUG[13050] netsock2.c: ...host '192.168.217.179' and port '5060'.
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.217.179:5060
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: = Looking for Call ID: ec87bd10-24943216@192.168.217.179 (Checking From) --From tag ee6fd0c82b6871beo0 --To-tag as3a28aa6e
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Stopping retransmission on 'ec87bd10-24943216@192.168.217.179' of Response 101: Match Found
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: = Looking for Call ID: ec87bd10-24943216@192.168.217.179 (Checking From) --From tag ee6fd0c82b6871beo0 --To-tag
[Dec 22 08:37:36] DEBUG[13050] netsock2.c: Splitting '77.220.130.136' gives...
[Dec 22 08:37:36] DEBUG[13050] netsock2.c: ...host '77.220.130.136' and port '(null)'.
[Dec 22 08:37:36] DEBUG[13050] netsock2.c: Splitting '77.220.130.136' gives...
[Dec 22 08:37:36] DEBUG[13050] netsock2.c: ...host '77.220.130.136' and port '(null)'.
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
[Dec 22 08:37:36] DEBUG[13050] netsock2.c: Splitting '192.168.217.179:5060' gives...
[Dec 22 08:37:36] DEBUG[13050] netsock2.c: ...host '192.168.217.179' and port '5060'.
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x76a6658'
[Dec 22 08:37:36] DEBUG[13050] res_rtp_asterisk.c: Allocated port 51406 for RTP instance '0x76a6658'
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: RTP instance '0x76a6658' is setup and ready to go
[Dec 22 08:37:36] DEBUG[13050] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x76a6658'
[Dec 22 08:37:36] VERBOSE[13050] netsock2.c: == Using SIP RTP CoS mark 5
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Setting NAT on RTP to On
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Processing session-level SDP o=- 33692007 33692007 IN IP4 192.168.217.179... UNSUPPORTED.
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Processing session-level SDP s=-... UNSUPPORTED.
[Dec 22 08:37:36] DEBUG[13050] netsock2.c: Splitting '192.168.217.179' gives...
[Dec 22 08:37:36] DEBUG[13050] netsock2.c: ...host '192.168.217.179' and port '(null)'.
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.217.179... OK.
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: Setting payload 0 based on m type on 0x7f5e513625e0
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: Setting payload 4 based on m type on 0x7f5e513625e0
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: Setting payload 8 based on m type on 0x7f5e513625e0
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: Setting payload 18 based on m type on 0x7f5e513625e0
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: Setting payload 97 based on m type on 0x7f5e513625e0
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: Setting payload 98 based on m type on 0x7f5e513625e0
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: Setting payload 101 based on m type on 0x7f5e513625e0
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000... OK.
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:2 G726-32/8000... OK.
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:4 G723/8000... OK.
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000... OK.
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:18 G729a/8000... OK.
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:96 G726-40/8000... OK.
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:97 G726-24/8000... OK.
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:98 G726-16/8000... OK.
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:100 NSE/8000... OK.
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Processing media-level (audio) SDP a=fmtp:100 192-193... UNSUPPORTED.
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000... OK.
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-15... UNSUPPORTED.
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Processing media-level (audio) SDP a=ptime:30... OK.
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Processing media-level (audio) SDP a=sendrecv... OK.
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: Incorporating payload 0 on 0x7f5e513625e0
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: Incorporating payload 2 on 0x7f5e513625e0
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: Incorporating payload 4 on 0x7f5e513625e0
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: Incorporating payload 8 on 0x7f5e513625e0
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: Incorporating payload 18 on 0x7f5e513625e0
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: Incorporating payload 97 on 0x7f5e513625e0
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: Incorporating payload 98 on 0x7f5e513625e0
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: Incorporating payload 101 on 0x7f5e513625e0
[Dec 22 08:37:36] DEBUG[13050] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x76a6658'
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: Copying payload 0 from 0x7f5e513625e0 to 0x76a6820
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: Copying payload 2 from 0x7f5e513625e0 to 0x76a6820
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: Copying payload 4 from 0x7f5e513625e0 to 0x76a6820
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: Copying payload 8 from 0x7f5e513625e0 to 0x76a6820
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: Copying payload 18 from 0x7f5e513625e0 to 0x76a6820
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: Copying payload 97 from 0x7f5e513625e0 to 0x76a6820
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: Copying payload 98 from 0x7f5e513625e0 to 0x76a6820
[Dec 22 08:37:36] DEBUG[13050] rtp_engine.c: Copying payload 101 from 0x7f5e513625e0 to 0x76a6820
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: We're settling with these formats: 0x100d0d (g723|ulaw|alaw|g726|g729|ilbc|h263p)
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Checking SIP call limits for device 7356322
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Updating call counter for incoming call
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: *** Our native formats are 0x100008 (alaw|h263p)
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: *** Joint capabilities are 0x100d0d (g723|ulaw|alaw|g726|g729|ilbc|h263p)
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: *** Our capabilities are 0x80030c7fffff (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)
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw)
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: This channel will not be able to handle video.
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: build_route: Contact hop: 7356322
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: SIP/7356322-00000014: New call is still down.... Trying...
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.217.179:5060
[Dec 22 08:37:36] DEBUG[13041] devicestate.c: No provider found, checking channel drivers for SIP - 7356322
[Dec 22 08:37:36] DEBUG[13041] chan_sip.c: Checking device state for peer 7356322
[Dec 22 08:37:36] DEBUG[13041] devicestate.c: Changing state for SIP/7356322 - state 1 (Not in use)
[Dec 22 08:37:36] DEBUG[13041] devicestate.c: device 'SIP/7356322' state '1'
[Dec 22 08:37:36] DEBUG[7656] pbx.c: Launching 'Dial'
[Dec 22 08:37:36] VERBOSE[7656] pbx.c: -- Executing [9002@users:1] Dial("SIP/7356322-00000014", "SIP/soberhome,20") in new stack
[Dec 22 08:37:36] DEBUG[7656] chan_sip.c: Asked to create a SIP channel with formats: 0x100008 (alaw|h263p)
[Dec 22 08:37:36] DEBUG[7656] chan_sip.c: Allocating new SIP dialog for 33d8d6ed43254c282d82e1b839a8c827@77.220.130.136:0 - INVITE (No RTP)
[Dec 22 08:37:36] DEBUG[7656] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x76cb888'
[Dec 22 08:37:36] DEBUG[7656] res_rtp_asterisk.c: Allocated port 51834 for RTP instance '0x76cb888'
[Dec 22 08:37:36] DEBUG[7656] rtp_engine.c: RTP instance '0x76cb888' is setup and ready to go
[Dec 22 08:37:36] DEBUG[7656] rtp_engine.c: Using engine 'asterisk' for RTP instance '0x76cf9f8'
[Dec 22 08:37:36] DEBUG[7656] res_rtp_asterisk.c: Allocated port 51160 for RTP instance '0x76cf9f8'
[Dec 22 08:37:36] DEBUG[7656] rtp_engine.c: RTP instance '0x76cf9f8' is setup and ready to go
[Dec 22 08:37:36] DEBUG[7656] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x76cf9f8'
[Dec 22 08:37:36] DEBUG[7656] res_rtp_asterisk.c: Setup RTCP on RTP instance '0x76cb888'
[Dec 22 08:37:36] VERBOSE[7656] netsock2.c: == Using SIP RTP CoS mark 5
[Dec 22 08:37:36] DEBUG[7656] chan_sip.c: Setting NAT on RTP to On
[Dec 22 08:37:36] DEBUG[7656] chan_sip.c: Setting NAT on VRTP to On
[Dec 22 08:37:36] DEBUG[7656] acl.c: For destination '192.168.217.182', our source address is '192.168.217.11'.
[Dec 22 08:37:36] DEBUG[7656] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.217.11:5060
[Dec 22 08:37:36] DEBUG[7656] chan_sip.c: *** Our native formats are 0x100008 (alaw|h263p)
[Dec 22 08:37:36] DEBUG[7656] chan_sip.c: *** Joint capabilities are 0x100008 (alaw|h263p)
[Dec 22 08:37:36] DEBUG[7656] chan_sip.c: *** Our capabilities are 0x80030c7fffff (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)
[Dec 22 08:37:36] DEBUG[7656] chan_sip.c: *** AST_CODEC_CHOOSE formats are 0x8 (alaw)
[Dec 22 08:37:36] DEBUG[7656] chan_sip.c: *** Our preferred formats from the incoming channel are 0x100008 (alaw|h263p)
[Dec 22 08:37:36] DEBUG[7656] chan_sip.c: This channel can handle video! HOLLYWOOD next!
[Dec 22 08:37:36] DEBUG[7656] rtp_engine.c: Seeded SDP of 'SIP/soberhome-00000015' with that of 'SIP/7356322-00000014'
[Dec 22 08:37:36] DEBUG[7656] channel.c: Not copying variable DIALEDTIME.
[Dec 22 08:37:36] DEBUG[7656] channel.c: Not copying variable ANSWEREDTIME.
[Dec 22 08:37:36] DEBUG[7656] channel.c: Not copying variable DIALEDPEERNAME.
[Dec 22 08:37:36] DEBUG[7656] channel.c: Not copying variable DIALEDPEERNUMBER.
[Dec 22 08:37:36] DEBUG[7656] channel.c: Not copying variable DIALSTATUS.
[Dec 22 08:37:36] DEBUG[7656] channel.c: Not copying variable SIPCALLID.
[Dec 22 08:37:36] DEBUG[7656] channel.c: Not copying variable SIPDOMAIN.
[Dec 22 08:37:36] DEBUG[7656] channel.c: Not copying variable SIPURI.
[Dec 22 08:37:36] DEBUG[7656] chan_sip.c: Outgoing Call for soberhome
[Dec 22 08:37:36] DEBUG[7656] chan_sip.c: Updating call counter for outgoing call
[Dec 22 08:37:36] DEBUG[7656] chan_sip.c: This call needs video offers!
[Dec 22 08:37:36] DEBUG[7656] chan_sip.c: This call needs text offers, but there's no text support enabled !
[Dec 22 08:37:36] DEBUG[7656] chan_sip.c: ** Our capability: 0x80020c109afe (gsm|ulaw|alaw|g726|adpcm|slin|lpc10|speex|speex16|g726aal2|g722|slin16|h263p|red|t140|testlaw) Video flag: False Text flag: False
[Dec 22 08:37:36] DEBUG[7656] chan_sip.c: ** Our prefcodec: 0x100008 (alaw|h263p)
[Dec 22 08:37:36] DEBUG[7656] chan_sip.c: -- Done with adding codecs to SDP
[Dec 22 08:37:36] DEBUG[7656] chan_sip.c: Done building SDP. Settling with this capability: 0x80020c109afe (gsm|ulaw|alaw|g726|adpcm|slin|lpc10|speex|speex16|g726aal2|g722|slin16|h263p|red|t140|testlaw)
[Dec 22 08:37:36] DEBUG[7656] chan_sip.c: Initializing initreq for method INVITE - callid 2a752ea219606acb35a6a55f19e10212@192.168.217.11:5060
[Dec 22 08:37:36] DEBUG[7656] chan_sip.c: Trying to put 'INVITE sip:' onto UDP socket destined for 192.168.217.182:5060
[Dec 22 08:37:36] VERBOSE[7656] app_dial.c: -- Called soberhome
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: = Looking for Call ID: 2a752ea219606acb35a6a55f19e10212@192.168.217.11:5060 (Checking To) --From tag as49f813c6 --To-tag
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '2a752ea219606acb35a6a55f19e10212@192.168.217.11:5060' Request 102: Found
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: SIP response 100 to standard invite
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: = Looking for Call ID: 2a752ea219606acb35a6a55f19e10212@192.168.217.11:5060 (Checking To) --From tag as49f813c6 --To-tag 326088956
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '2a752ea219606acb35a6a55f19e10212@192.168.217.11:5060' Request 102: Found
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: SIP response 101 to standard invite
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: = Looking for Call ID: 1897562645 (Checking From) --From tag 570230773 --To-tag as49f813c6
[Dec 22 08:37:36] DEBUG[13050] acl.c: For destination '192.168.217.182', our source address is '192.168.217.11'.
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Setting SIP_TRANSPORT_UDP with address 192.168.217.11:5060
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Allocating new SIP dialog for 1897562645 - OPTIONS (No RTP)
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: **** Received OPTIONS (3) - Command in SIP OPTIONS
[Dec 22 08:37:36] DEBUG[13050] chan_sip.c: Trying to put 'SIP/2.0 481' onto UDP socket destined for 192.168.217.182:5060
[Dec 22 08:37:37] DEBUG[13050] chan_sip.c: = Looking for Call ID: 2a752ea219606acb35a6a55f19e10212@192.168.217.11:5060 (Checking To) --From tag as49f813c6 --To-tag 326088956
[Dec 22 08:37:37] DEBUG[13050] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '2a752ea219606acb35a6a55f19e10212@192.168.217.11:5060' Request 102: Found
[Dec 22 08:37:37] DEBUG[13050] chan_sip.c: SIP response 180 to standard invite
[Dec 22 08:37:37] DEBUG[13041] devicestate.c: No provider found, checking channel drivers for SIP - soberhome
[Dec 22 08:37:37] DEBUG[13041] chan_sip.c: Checking device state for peer soberhome
[Dec 22 08:37:37] DEBUG[13041] devicestate.c: Changing state for SIP/soberhome - state 1 (Not in use)
[Dec 22 08:37:37] DEBUG[13041] devicestate.c: device 'SIP/soberhome' state '1'
[Dec 22 08:37:37] VERBOSE[7656] app_dial.c: -- SIP/soberhome-00000015 is ringing
[Dec 22 08:37:37] DEBUG[7656] rtp_engine.c: Setting early bridge SDP of 'SIP/7356322-00000014' with that of 'SIP/soberhome-00000015'
[Dec 22 08:37:37] DEBUG[7656] chan_sip.c: Trying to put 'SIP/2.0 180' onto UDP socket destined for 192.168.217.179:5060
[Dec 22 08:37:38] DEBUG[13050] chan_sip.c: = Looking for Call ID: 34d21bfe-d2229876@192.168.217.179 (Checking From) --From tag f6a686de1d6e2b16o0 --To-tag
[Dec 22 08:37:38] DEBUG[13050] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY
[Dec 22 08:37:38] DEBUG[13050] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.217.179:5060
[Dec 22 08:37:43] DEBUG[13050] chan_sip.c: = Looking for Call ID: 34d21bfe-d2229876@192.168.217.179 (Checking From) --From tag f6a686de1d6e2b16o0 --To-tag
[Dec 22 08:37:43] DEBUG[13050] chan_sip.c: **** Received NOTIFY (4) - Command in SIP NOTIFY
[Dec 22 08:37:43] DEBUG[13050] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.217.179:5060
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: = Looking for Call ID: 2a752ea219606acb35a6a55f19e10212@192.168.217.11:5060 (Checking To) --From tag as49f813c6 --To-tag 326088956
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: Acked pending invite 102
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: Stopping retransmission on '2a752ea219606acb35a6a55f19e10212@192.168.217.11:5060' of Request 102: Match Found
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: SIP response 200 to standard invite
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: Processing session-level SDP v=0... UNSUPPORTED.
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: Processing session-level SDP o=sober 123456 654321 IN IP4 192.168.217.182... UNSUPPORTED.
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: Processing session-level SDP s=A conversation... UNSUPPORTED.
[Dec 22 08:37:45] DEBUG[13050] netsock2.c: Splitting '192.168.217.182' gives...
[Dec 22 08:37:45] DEBUG[13050] netsock2.c: ...host '192.168.217.182' and port '(null)'.
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: Processing session-level SDP c=IN IP4 192.168.217.182... OK.
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: Processing session-level SDP t=0 0... UNSUPPORTED.
[Dec 22 08:37:45] DEBUG[13050] rtp_engine.c: Setting payload 8 based on m type on 0x7f5e51361b80
[Dec 22 08:37:45] DEBUG[13050] rtp_engine.c: Setting payload 0 based on m type on 0x7f5e51361b80
[Dec 22 08:37:45] DEBUG[13050] rtp_engine.c: Setting payload 110 based on m type on 0x7f5e51361b80
[Dec 22 08:37:45] DEBUG[13050] rtp_engine.c: Setting payload 117 based on m type on 0x7f5e51361b80
[Dec 22 08:37:45] DEBUG[13050] rtp_engine.c: Setting payload 101 based on m type on 0x7f5e51361b80
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:8 PCMA/8000/1... OK.
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:0 PCMU/8000/1... OK.
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:110 speex/8000/1... OK.
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: Processing media-level (audio) SDP a=fmtp:110 vbr=on... UNSUPPORTED.
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:117 speex/16000/1... OK.
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: Processing media-level (audio) SDP a=fmtp:117 vbr=on... UNSUPPORTED.
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/8000/1... OK.
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: Processing media-level (audio) SDP a=fmtp:101 0-11... UNSUPPORTED.
[Dec 22 08:37:45] DEBUG[13050] rtp_engine.c: Setting payload 98 based on m type on 0x7f5e51360b00
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: Processing media-level (video) SDP a=rtpmap:98 H263-1998/90000... OK.
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: Processing media-level (video) SDP a=fmtp:98 CIF=1;QCIF=1... UNSUPPORTED.
[Dec 22 08:37:45] DEBUG[13050] rtp_engine.c: Incorporating payload 0 on 0x7f5e51361b80
[Dec 22 08:37:45] DEBUG[13050] rtp_engine.c: Incorporating payload 8 on 0x7f5e51361b80
[Dec 22 08:37:45] DEBUG[13050] rtp_engine.c: Incorporating payload 101 on 0x7f5e51361b80
[Dec 22 08:37:45] DEBUG[13050] rtp_engine.c: Incorporating payload 110 on 0x7f5e51361b80
[Dec 22 08:37:45] DEBUG[13050] rtp_engine.c: Incorporating payload 117 on 0x7f5e51361b80
[Dec 22 08:37:45] DEBUG[13050] rtp_engine.c: Incorporating payload 98 on 0x7f5e51360b00
[Dec 22 08:37:45] DEBUG[13050] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x76cb888'
[Dec 22 08:37:45] DEBUG[13050] rtp_engine.c: Copying payload 0 from 0x7f5e51361b80 to 0x76cba50
[Dec 22 08:37:45] DEBUG[13050] rtp_engine.c: Copying payload 8 from 0x7f5e51361b80 to 0x76cba50
[Dec 22 08:37:45] DEBUG[13050] rtp_engine.c: Copying payload 101 from 0x7f5e51361b80 to 0x76cba50
[Dec 22 08:37:45] DEBUG[13050] rtp_engine.c: Copying payload 110 from 0x7f5e51361b80 to 0x76cba50
[Dec 22 08:37:45] DEBUG[13050] rtp_engine.c: Copying payload 117 from 0x7f5e51361b80 to 0x76cba50
[Dec 22 08:37:45] DEBUG[13050] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x76cf9f8'
[Dec 22 08:37:45] DEBUG[13050] rtp_engine.c: Copying payload 98 from 0x7f5e51360b00 to 0x76cfbc0
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: We're settling with these formats: 0x20010020c (ulaw|alaw|speex|speex16|h263p)
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: We have an owner, now see if we need to change this call
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: Updating call counter for outgoing call
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: build_route: Contact hop:
[Dec 22 08:37:45] DEBUG[13050] netsock2.c: Splitting '192.168.217.182' gives...
[Dec 22 08:37:45] DEBUG[13050] netsock2.c: ...host '192.168.217.182' and port '(null)'.
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: Trying to put 'ACK sip:sob' onto UDP socket destined for 192.168.217.182:5060
[Dec 22 08:37:45] DEBUG[13041] devicestate.c: No provider found, checking channel drivers for SIP - soberhome
[Dec 22 08:37:45] DEBUG[13041] chan_sip.c: Checking device state for peer soberhome
[Dec 22 08:37:45] VERBOSE[7656] app_dial.c: -- SIP/soberhome-00000015 answered SIP/7356322-00000014
[Dec 22 08:37:45] DEBUG[13041] devicestate.c: Changing state for SIP/soberhome - state 1 (Not in use)
[Dec 22 08:37:45] DEBUG[13041] devicestate.c: device 'SIP/soberhome' state '1'
[Dec 22 08:37:45] DEBUG[7656] rtp_engine.c: Setting early bridge SDP of 'SIP/7356322-00000014' with that of 'SIP/soberhome-00000015'
[Dec 22 08:37:45] DEBUG[13041] devicestate.c: No provider found, checking channel drivers for SIP - 7356322
[Dec 22 08:37:45] DEBUG[7656] chan_sip.c: SIP answering channel: SIP/7356322-00000014
[Dec 22 08:37:45] DEBUG[13041] chan_sip.c: Checking device state for peer 7356322
[Dec 22 08:37:45] DEBUG[7656] res_rtp_asterisk.c: Setting the marker bit due to a source update
[Dec 22 08:37:45] DEBUG[13041] devicestate.c: Changing state for SIP/7356322 - state 1 (Not in use)
[Dec 22 08:37:45] DEBUG[13041] devicestate.c: device 'SIP/7356322' state '1'
[Dec 22 08:37:45] DEBUG[7656] chan_sip.c: Setting framing from config on incoming call
[Dec 22 08:37:45] DEBUG[7656] chan_sip.c: ** Our capability: 0x100d0d (g723|ulaw|alaw|g726|g729|ilbc|h263p) Video flag: True Text flag: True
[Dec 22 08:37:45] DEBUG[7656] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
[Dec 22 08:37:45] DEBUG[7656] chan_sip.c: -- Done with adding codecs to SDP
[Dec 22 08:37:45] DEBUG[7656] chan_sip.c: Done building SDP. Settling with this capability: 0x100d0d (g723|ulaw|alaw|g726|g729|ilbc|h263p)
[Dec 22 08:37:45] DEBUG[7656] chan_sip.c: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.217.179:5060
[Dec 22 08:37:45] DEBUG[7656] features.c: bridge answer set, chan answer set
[Dec 22 08:37:45] DEBUG[7656] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet
[Dec 22 08:37:45] DEBUG[7656] res_rtp_asterisk.c: Not changing SSRC since we haven't sent any RTP yet
[Dec 22 08:37:45] VERBOSE[7656] rtp_engine.c: -- Locally bridging SIP/7356322-00000014 and SIP/soberhome-00000015
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: = Looking for Call ID: ec87bd10-24943216@192.168.217.179 (Checking From) --From tag ee6fd0c82b6871beo0 --To-tag as6fb727c9
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: Stopping retransmission on 'ec87bd10-24943216@192.168.217.179' of Response 102: Match Found
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ec87bd10-24943216@192.168.217.179' Method: ACK
[Dec 22 08:37:45] DEBUG[13050] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2a752ea219606acb35a6a55f19e10212@192.168.217.11:5060' Method: INVITE
[Dec 22 08:37:46] DEBUG[13050] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ec87bd10-24943216@192.168.217.179' Method: ACK
[Dec 22 08:37:46] DEBUG[13050] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2a752ea219606acb35a6a55f19e10212@192.168.217.11:5060' Method: INVITE
[Dec 22 08:37:47] DEBUG[13050] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ec87bd10-24943216@192.168.217.179' Method: ACK
[Dec 22 08:37:47] DEBUG[13050] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog '2a752ea219606acb35a6a55f19e10212@192.168.217.11:5060' Method: INVITE
[Dec 22 08:37:48] DEBUG[13050] chan_sip.c: Bridge still active. Delaying destroy of SIP dialog 'ec87bd10-24943216@192.168.217.179' Method: ACK