На 03.02.11 21:03, John Ervin написа:
I use SIP Phones in general a lot to make direct SIP calls. I don't
need to have a host to make one of these calls, just specify something
like sip:firstname.lastname@example.org to connect to the VOIP Users Conference
on Fridays at noon (You can call in too if you like)... It appears to
me that SIP Communicator wants me to set up an account that SC will
register to and then contact through the host.
Although, for a variety of reasons, this would definitely be the best
way to connect, SC does not require you to use a server. In order to
make calls without using a proxy you need to create what we call a
Registrarless account. To do so, create a new SIP account and only enter
a user name without a server part. You should see an example in the
account wizard. In other words, if you are determined to run without a
server then instead of setting your account as "email@example.com" you
should configure it as simply "jervin".
b) When I make the ZipDX call each friday with SC, it works for about
2 to 3 minutes and then the call drops.
Strange. I often connect to VUC (using SC of course) and never had the
issue. My last call was more than a month ago however so something may
have changed since then.
One thing we changed recently and that may be related is the fact that,
for security reasons, we started dropping requests that arrive from a
server that we are not connected to. Of course we only do this for
accounts that actually have servers.
I contacted ZipDX about this
(David Frankelfirstname.lastname@example.org / 1-800-372-6535). He says _"I looked
at our logs. Your client is not responding to the SIP UPDATE messages
that we send periodically. Even if a client doesn't support a
particular method, it still needs to respond (with, for example, "405
METHOD NOT ALLOWED"). The client's failure to respond (after many
retries) makes us assume that it has gone away, so we disconnect."_
I just had a call with email@example.com and got dropped exactly 10
minutes after I started the call. I am not sure whether this was the
same problem as the one you encountered since it could just be about
how wbdemo is configured.
However, I didn't receive a single SIP packet from zipdx between the
initial INVITE transaction and the BYE request. In other words there was
nothing for us to ignore.
I won't be able to join tomorrow's VUC because of FOSDEM so if you or
anyone else could be there and sees the problem, then it would be nice
to see the logs (and if posible pcap dumps).
Item B makes SIP Communicator almost unusable (have to reconnect every 2
to 3 minutes) in this setting. I'm assuming that the ZipDX people deal
with a lot of SIP dialers and know what they are talking about, but I've
put the ZipDX contact persons info in the message in case you want to
John F. Ervin
Central Florida TeleSource
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
firstname.lastname@example.org PHONE: +220.127.116.11.43.30
http://sip-communicator.org FAX: +18.104.22.168.47.31