SIP videosupport

When I enter “videosupport=yes” in sip.conf I can’t call any SIP phone on my * (all of them are “on the phone” - and I directly go to voicemail). When I comment this line (;), everything works normal. Why * act this way? For videocalls I use eyeBeam.

Thank you for your time!

My sip.conf

[general]
externip = 111.222.333.444
fromdomain=mydomain.hr
localnet=10.0.0.0/255.255.255.0
port=5060
bindaddr=0.0.0.0
context=sip
srvlookup=yes
dtmfmode=rfc2833
disallow=all
allow=gsm
allow=ulaw
allow=alaw
musicclass=default
videosupport=yes

[2026]
type=friend
username=2026
secret=2026
host=dynamic
mailbox=2026
callerid=First Last <2026>
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=h263
allow=h263p
allow=h261
canreinvite=no

[2031]
type=friend
username=2031
secret=2031
host=dynamic
mailbox=2031
callerid=First Last <2031>
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=h263
allow=h263p
allow=h261
canreinvite=no

u need to specify the video codec in the [general] section as well…

allow=h263

[quote=“mtyoung”]u need to specify the video codec in the [general] section as well…

allow=h263[/quote]

I have tried. It doesn’t help.
Why asterisk thinks that phone is in use when it isn’t?! I’m going crazy!!!

It can’t be that I’m the only one with this situation!?!

Parcina, I’m not clear on exactly what your problem is but it looks like you are just using the default dialplan for all your users…context=SIP…Try just putting a specific context in the extensions.conf for your users.

My problem is when I add this line “videosupport=yes” in sip.conf. Every phone that is registred with Asterisk, when someone trys to call it, is “on the phone”.

I have two soft phones on two computers (one next to other). And when I from first softphone call second, my call directly goes to voiceamill (asterisk says that other softphone is on the phone).

When I delete that line “videosupport=yes” everything works fine (except I don’t have video :wink:)

parcina, thats certainly very odd…haven’t come across that one!

I defenetly agree with you. Until I haven’t try it several times I didn’t belive it myself. And I change only that one line and restart Asterisk…

Hi,
Even I encountered with that type of a problem too…
I try to use asterisk with public IP with my dsl link

Settings

in sip.conf settings, (global)

videosupport=yes

in each account at sip.conf you have to define

allow=ulaw
allow=alaw
allow=gsm
allow=h263
allow=h263p


my problem is once this worked fine. next day when i want to demonstrate this it didint work… I think asterisk is experencing some NAT problems. If your sip clients are behind a NAT then he cant find his path through the NAT…extensions are registering fine…but we cant call a number.

Is anyone out there to share some info about video conf. just hit me on skype id hacksics

:smile:

Hi, I came across this thread searching for video information.

I’ve got a similar problem using a pair of Polycom VSX300 videoconference stations. Did anyone work out what was happening here?

Thanks

PJH

im having a very similar problem. when videosupport=yes in sip.conf, only one person can use a phone at any one time and the rest all go to voicemail. any ideas anyone??

cheers