Как заставить asterisk передавать видео при звонке?

[general]
alwaysauthreject=yes
allowguest=no
language=ru
pedantic=no
nat=force_rport,comedia
rtptimeout=10
externtcpport=5060
port=5060
disallow=all
canreinvite=no
defaultexpiry=120
videosupport=yes
textsupport=yes

[phones](!)
type=friend
context=phones
host=dynamic
nat=yes
qualify=yes
canreinvite=no
disallow=all
allow=alaw
allow=ulaw
;allow=h261
;allow=h263
;allow=h263p
allow=h264
;allow=vp8
;allow=mpeg4
dtmfmode=auto
callcounter=yes
call-limit=10

[100](phones)
callerid="100" <100>
mailbox=100@default

Sending to 192.168.1.79:46682 (NAT)
Using INVITE request as basis request - -cYIvjejgXdKRAZZwmxRPw..
Found peer '100' for '100' from 192.168.1.79:46682
  == Using SIP VIDEO CoS mark 6
  == Using SIP RTP CoS mark 5
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|ulaw|h264), peer - audio=(ulaw|gsm|alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.79:54946
Peer doesn't provide video
Peer doesn't provide T.140
Looking for 101 in phones (domain 192.168.1.130)
sip_route_dump: route/path hop: <sip:100@192.168.1.79:46682;transport=UDP>