Прописал все кодеки,

in your sip.conf:

[general]
videosupport=yes ; enable Asterisk video support

[101]
type=friend
username=101
secret=hidden
host=dynamic
context=internal
callerid=Video Phone <101>
disallow=all ; better for custom-tunning codec selection
allow=ulaw
allow=alaw
allow=gsm
allow=h263 ; H.263 is our video codec
allow=h264
allow=h263p ; H.263p is the enhanced video codec
dtmfmode=rfc2833 ; inband is not supported in compressed codecs like gsm, so we better set it to rfc2833
canreinvite=no ; canreinvite must be set to 'no'

Но нет видео,

[Dec 24 16:20:42] NOTICE[25252]: rtp.c:1286 ast_rtp_read: Unknown RTP codec 126 received from '0.0.0.0'

Попробуй оставить один видео кодек разрешенный.

Оставил только allow=h263p, но также
[Dec 24 16:20:42] NOTICE[25252]: rtp.c:1286 ast_rtp_read: Unknown RTP codec 126 received from '0.0.0.0'

Ошибка относится не к видео. ТАМ говорят, что

It's an X-Lite problem that they did not first negotiate codec 126 in the SIP dialog before sending it. It is explicitly a violation of the SIP specification to send an non-negotiated codec, and we WANT to know when that happens, so we can inform vendors to fix their clients.

Это что касается payload'a 126.

damir_t , * какой?

На передачу видео по h263+ это не влияло, только на консоль мусорило.

asterisk -rx 'core show version'
Asterisk 1.4.21.2 built by root @ Asterisk on a i686 running Linux on 2009-10-20 22:43:52 UTC

debug при звонке с Saryagash_1 на video, клиенты Bria:

<--- SIP read from 192.168.1.24:19048 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK4a0bcbf8;rport=5060
Contact: <sip:video@192.168.1.24:19048;rinstance=19a8bd846a338267>
To: <sip:video@192.168.1.24:19048;rinstance=19a8bd846a338267>;tag=60736b22
From: "Saryagash_1"<sip:7253701@192.168.1.9>;tag=as365332fb
Call-ID: 695c75e948d94ef27b85744e5c2c518d@192.168.1.9
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: Bria release 2.5 RC4 stamp 47242
Content-Length: 402

v=0
o=- 9 2 IN IP4 192.168.1.24
s=CounterPath Bria
c=IN IP4 192.168.1.24
t=0 0
m=audio 28728 RTP/AVP 0 8 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:E72081D7E6974B248E9B14C214750C8F
m=video 42070 RTP/AVP 103
a=fmtp:103 QCIF=1 CIF=1 I=1 J=1 T=1 MaxBR=4520
a=inactive
a=rtpmap:103 H263-1998/90000
a=x-rtp-session-id:9BD66AC16ED04B8F96174CE69901FDE9

<------------->
--- (11 headers 15 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found RTP video format 103
Peer audio RTP is at port 192.168.1.24:28728
Found audio description format telephone-event for ID 101
Found video description format H263-1998 for ID 103
Capabilities: us - 0x14000e (gsm|ulaw|alaw|h261|h263p), peer - audio=0x10000c (ulaw|alaw|h263p)/video=0x100000 (h263p), combined - 0x10000c (ulaw|alaw|h263p)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.24:28728
Peer video RTP is at port 192.168.1.24:42070
list_route: hop: <sip:video@192.168.1.24:19048;rinstance=19a8bd846a338267>
set_destination: Parsing <sip:video@192.168.1.24:19048;rinstance=19a8bd846a338267> for address/port to send to
set_destination: set destination to 192.168.1.24, port 19048
Transmitting (no NAT) to 192.168.1.24:19048:
ACK sip:video@192.168.1.24:19048;rinstance=19a8bd846a338267 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9:5060;branch=z9hG4bK24413db8;rport
From: "Saryagash_1" <sip:7253701@192.168.1.9>;tag=as365332fb
To: <sip:video@192.168.1.24:19048;rinstance=19a8bd846a338267>;tag=60736b22
Contact: <sip:7253701@192.168.1.9>
Call-ID: 695c75e948d94ef27b85744e5c2c518d@192.168.1.9
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Saryagash_1" <sip:7253701@192.168.1.9>;privacy=off;screen=no
Content-Length: 0

RTP бежит после установления сессии? Звук есть? Напрямую видео ходит?