We’ve just got a VPS with asterisk 1.8.13.1 in the repo.
I’m having a bit of a problem with video conferencing. I can’t find any relevant docs that go into confbridge in enough detail for video for that version.
I’m also not sure of what ports should be forwarded on the router of the client machines for video support. Searching the net seems to find various port lists. In fact, I couldn’t even get video working for the latest version 11.8.1.
Any info/help hugely appreciated, as always.
-Dave
*Edit: here are the conf files that I’m using:
confbridge.conf:
[code][general]
[default_bridge]
type=bridge
max_members=20
mixing_interval=10
internal_sample_rate=auto
record_conference=no
video_mode=follow_talker
[default_user]
type=user
announce_user_count_all=yes
announce_join_leave=yes
dsp_drop_silence=yes
denoise=yes
pin=5555 ; Pin doesn’t work.[/code]
extensions.conf:
[code][general]
static=yes
writeprotect=yes
autofallthrough=yes
[incoming]
exten => 1234,1,Answer()
exten => 1234,n,GotoIf($["${CALLERID(name)}" = “MyNick”]?admin)
exten => 1234,n,ConfBridge(1234, 5555) ; Pin doesn’t work here either
exten => 1234,n,Hangup()
exten => 1234,n(admin),ConfBridge(1234, a)
exten => 1234,n,Hangup()[/code]
sip.conf:
[general]
alwaysauthreject=yes
canreinvite=yes
Qualify=yes
allowguest=yes
context=incoming
allowsubscribe=yes
dtmfmode=auto
allowoverlap=no
bindport=40000
udpbindaddr=0.0.0.0
transport=udp
srvlookup=yes
limitonpeers=no
videosupport=yes
textsupport=yes
callevents=yes
notifyringing=yes
notifyhold=yes
registertimeout=60
limitonpeers=yes
call-limit=100
localnet=192.168.1.0/255.255.255.0
externhost=our.domain.com
disallow=all
allow=alaw
allow=ulaw
allow=gsm
allow=speex
I’ve come across a dialplan that uses a manually entered PIN using read, but that’s not really a problem. It’s the video I’d like to get working more than anything.
*edit 2:
Added to sip.conf [general]:
allow=h261
allow=h263
allow=h263p
Now I can see local video at least.