Pollock, Chase wrote:
First of all, thank you for the reply, it seems to be full of great
information.
I apologize for the lack of description, I do that occasionally(Know
exactly what is wrong in my head so I don't think other people don't
know, hah). Anyway, Before I received your response I played with my
sip.conf and I added Autocreatepeer=yes, insecure=port,invite. I also
changed my nat=route to nat=yes.
I haven't used any 'insecure' settings for any of my openwengo clients
(5 so far). they all login fine. I *do* use 'nat=yes' because openwengo
won't do STUN, so we have to ask asterisk to try to fixup the NAT
issues. if your clients are all on the same LAN, you should use
'nat=never', and you *might* be able to use 'canreinvite=yes' too.
I don't use autocreatepeer=yes. try hard coding some 'friend' devices.
With these settings, 3 of the test machines seem to be able to connect
fine to the Asterisk server, and *place* calls between each other. The
weird thing is, there was no ringing, at which point I added the
progressinband=yes as you suggested, but still no ringing.
that's odd, but I think related to your later statement that audio
doesn't work. the 'progressinband=yes' flag just tells asterisk to
answer the call immediately and make ringing noises for you, since the
SIP device (openwengo in this case) can't make them itself.
Additionally, once the call is placed on one end, it takes about 5
seconds for the notification to show up on the other computer, is this
normal. For this, I changed my canreinvite=no to yes, at which point
nope. mine ring instantly.
you might try killing and restarting the openwengo clients after you
make asterisk config changes, openwengo is not terribly stable when the
config options are wrong, it tends to hang, take long pauses, or crash...
the calls went straight through about half the time, and delayed several
seconds(or not at all) the other half. I don't know if the
canreinvite=yes is the cause of this, but sometimes the call doesn't
even go through.
openwengo doesn't cope well with canreinvite=yes, certainly in the NAT
situation because of lack of STUN. it might work for LAN, but I'd leave
it out to start with because it complicates things. basically it means
that (1) the clients start talking to one another via asterisk, then (2)
a few seconds into the conversation, asterisk tries to make the clients
talk directly to one another.
with canreinvite=no, they just continue talking to one another via asterisk.
Another issue, once the notification shows up, and call answered, there
is no sound transfer between the two phones. I'm not sure if this is a
mic issue or not, but the mic does work(ie, with Ekiga). In the audio
settings I have Alsa selected for Input device.
hmm. all my openwengo clients are Windows. I haven't yet tried to
introduce linux into the mix... ALSA and friends have always been the
stumbling block for my attempts at VOIP on linux...
I added videosupport=yes in the event that I may use at some point, I'll
go fancy once I get basic going :)
try disabling videosupport. there's some comments around that openwengo
has problems if it can't get the webcam working.
For the supported codec's, I removed allow=gsm and added the h263/p.
the lack of audio makes me think it's attempting to use a codec it
doesn't really understand. I would try:
disallow=all
allow=ulaw
and see if that helps. alternatively, it's firewall related. each client
is sending audio to asterisk, which in turn sends it on to the other
client. you need to have 'related' traffic allowed.
the fact that ekiga works would suggest either you've already opened a
hole in the firewall for its RTP, or it's not the firewall ;-)
I'm not sure what you mean about presence? This might be related...but
If any of my users have passwords, they are unable to login.
OK, now we're getting really strange. I haven't had any issues with
logging in. try setting both the server and proxy fields to your server
name for starters. you should *absolutely* be able to login. try
different 'nat=' settings. also, check firewalls on your client machines.
also, are you running multiple SIP clients on the same machine? ie, same
time as openwengo? it doesn't play terribly well with other software
listening on the same port (5060). you can change that in the advanced
options, just pick something else like 5070...
The only thing that I did not add was the directrtpsetup=yes. Not sure
I understand what this does, maybe it's the reason for the above issues?
disable it if you've got it set. it absolutely won't work if you have
nat=yes set. basically if you have 'canreinvite=yes' and
'directrtpsetup=yes', asterisk will try to directly connect the clients
when they call one another, rather than the two step process I described
above. it will only work for public IP addresses, wide open NATs, or
LAN. it would be a nice to have, but not until you get basic stuff working.
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