[asterisk-dev] PJSIP transport problem
Steve Murphy
murf at parsetree.com
Fri Feb 28 12:58:34 CST 2014
On Fri, Feb 28, 2014 at 7:28 AM, Joshua Colp <jcolp at digium.com> wrote:
> On 14-02-28 10:24 AM, Steve Murphy wrote:
> > Josh--
> >
> > Thanks for that info... but...
> > can't I use different ports? (can't apache listen
> > on both port 80 and 443 on the same ipaddr?) I tried using
> > different ports and still have this problem. What do you advise
> > if I have some natted phones registering and some
> > not? I have some more questions on the subject of natting,
> > but I'll pose those in another thread.
>
> Yes, you can run them on different ports. What message are you getting?
>
Eh, after looking at this, I was changing the :5060 on
the external_signaling_port, and not on the bind, so
my bad, being a newby and all.
>
> As for your comment about natted phones the configuration options on
> transports control things when Asterisk is behind NAT. They specify what
> is local to it so SIP messages don't get rewritten, or if going outside
> of that - they do get rewritten to the external address.
>
> The options used when a remote endpoint is behind NAT are:
>
> force_rport=yes
> rtp_symmetric=yes
> rewrite_contact=yes
>
Many Thanks, Josh...
I removed the NAT options (external_signalling/media_*) set on transport,
and used the above suggested options on the endpoint,
and got it so phones can call other phones.
Two issues:
1.I notice that I can't see the registration status
of the various endpoints; is this something on the to-do
list of things to be developed, or am I missing seomthing?
I can get a list of outgoing registrations, tho.
2. I'm seeing nat table expirations drop out from under dumber
phones, where I can neither shorten the registration times
nor send options. Is there a way to get pjsip to send out
keepalives (OPTIONS)? It's kind of a bummer to have phones
that think they are still registered, but you can't reach them.
Looked thru the literature, haven't seen anything "juicy" about
this.
murf
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
>
--
Steve Murphy
ParseTree Corporation
57 Lane 17
Cody, WY 82414
✉ murf at parsetree dot com
☎ 307-899-5535307-899-5535
Call
Send SMS
Add to Skype
You'll need Skype CreditFree via Skype
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20140228/2d0cd4a4/attachment.html>
More information about the asterisk-dev
mailing list