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Posts: 3 | Thanked: 0 times | Joined on Aug 2010
#13
Originally Posted by jolouis View Post
Okay, first important step that you've got there is to make sure that in the global context "videosupport=yes" is set. You need to make sure that you have the right codecs enabled for each user though, as the users over-ride the global settings context.

So for example, in your users.conf file if you were extension 6000, you'd have an entry like this:
Code:
[6000]
username = 6000
transfer = yes
mailbox = 6000
call-limit = 100
fullname = Tablet
registersip = no
host = dynamic
callgroup = 1
context = DLPN_DialPlan1
cid_number = 6000
hasvoicemail = yes
vmsecret = 1234
email =
threewaycalling = no
hasdirectory = no
callwaiting = no
hasmanager = no
hasagent = no
hassip = yes
hasiax = yes
secret = 1234
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = no
pickupgroup = 1
autoprov = no
label =
macaddress =
linenumber = 1
disallow = all
allow = g729,gsm,ulaw,alaw,h263
Notice the last two lines: disallow = all, which is there by default. Then allow, which overrides it, specifying which codecs this user can support; the key for video is h263.
You also need to make sure that your asterisk was compiled with H263 support. To do this, connect to your asterisk box:

asterisk -r

then enter the command:
core show codecs

And you should get a list of all the codecs your build supports:
Code:
Disclaimer: this command is for informational purposes only.
        It does not indicate anything about your configuration.
        INT    BINARY        HEX   TYPE       NAME   DESC
--------------------------------------------------------------------------------
          1 (1 <<  0)      (0x1)  audio       g723   (G.723.1)
          2 (1 <<  1)      (0x2)  audio        gsm   (GSM)
          4 (1 <<  2)      (0x4)  audio       ulaw   (G.711 u-law)
          8 (1 <<  3)      (0x8)  audio       alaw   (G.711 A-law)
         16 (1 <<  4)     (0x10)  audio   g726aal2   (G.726 AAL2)
         32 (1 <<  5)     (0x20)  audio      adpcm   (ADPCM)
         64 (1 <<  6)     (0x40)  audio       slin   (16 bit Signed Linear PCM)
        128 (1 <<  7)     (0x80)  audio      lpc10   (LPC10)
        256 (1 <<  8)    (0x100)  audio       g729   (G.729A)
        512 (1 <<  9)    (0x200)  audio      speex   (SpeeX)
       1024 (1 << 10)    (0x400)  audio       ilbc   (iLBC)
       2048 (1 << 11)    (0x800)  audio       g726   (G.726 RFC3551)
       4096 (1 << 12)   (0x1000)  audio       g722   (G722)
      65536 (1 << 16)  (0x10000)  image       jpeg   (JPEG image)
     131072 (1 << 17)  (0x20000)  image        png   (PNG image)
     262144 (1 << 18)  (0x40000)  video       h261   (H.261 Video)
     524288 (1 << 19)  (0x80000)  video       h263   (H.263 Video)
    1048576 (1 << 20) (0x100000)  video      h263p   (H.263+ Video)
    2097152 (1 << 21) (0x200000)  video       h264   (H.264 Video)
H263 is the important one for video on the tablets.

Hope that helps!

Thanks,
-Rob

hi, i am kuki and i need your help about enabling video in asterisk.
here is my sip.conf:

[general]
videosupport=yes

[1000]
type=friend
context=incoming
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=h263
dtmfmode=rfc2833
canreinvite=no

[1003]
type=friend
context=incoming
host=dynamic
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=h263
allow=h263p
dtmfmode=rfc2833
canreinvite=no

and my extensions.conf is:

[globals]
leif=SIP/1003
kuki2=SIP/1000

[incoming]

exten => 1003,1,Dial(${leif},20)
exten => 1003,n,Playback(vm-nobodyavail)
exten => 1003,n,Hangup()

exten => 1000,1,Dial(${kuki2},20)
exten => 1000,n,Playback(vm-nobodyavail)
exten => 1000,n,Hangup()



i can send/receive voice between 1000 and 1003, but when i push send video button nothing happend. My softphone is x-lite 4.0 beta2 (i download it from here http://www.brothersoft.com/x-lite-95889.html)

My asterisk is asterisk-1.4.33.1

please if you have some idea help me

thank you in advance