asterisk users - Jun 2010 - Still sipping frustration - only getting state ACK

Hello everyone!
   I still am not much further along with my sip calling. I changed my sip.conf 
taking suggestions from the net (voip-info.org in particular). I changed 
iptel's position from friend to peer. I turned on and off nat, I chose 
different codecs in first place, entered my outward IP as fromdomain and 
uncommented the register directive with correct values.
   All I get is two registrations now, but no calls.  get a registration effort 
every 225secs and it succeeds. But when I make a call;
channel originate sip/iptel-out/echo at iptel.org Application playback 
vm/net_ring
   The call is onlyleft in state ACK for a while. Then asterisk tells me, that 
it is destroying the sip dialog (long ID) INVITE.
   Question: Might it be a problem, that my system only knows itself as 
192.168.*. Do I need to set something else than externip?
   Might it be, that my router really blocks certain ports? I can't check
it,
since it's heavily javascript based and, since I'm blind and the
accessibility
software for the GUI never really worked on this distro, I don't have a 
browser to look at it.
   Do I need to forward port 5060 to my machine specifically (like it is needed 
for SSH's port 22), or is the mechanism based on: I talk first and the sever
gets back to me based on that.
   This configuration worked for googletalk. I admit, there were problems, but 
calls were coming through from both sides.
   Please can someone help me clear up this mess. I'm completely frustrated
and
don't know what else to do, where else to look.
   Kindly yours and thanks in advance
             JUlien

--------
Music was my first love and it will be my last (John Miles)

======== FIND MY WEB-PROJECT AT: =======http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: ======http://www.juliencoder.de

Julien Claassen wrote:
> Hello everyone!
>    I still am not much further along with my sip calling. I changed my
sip.conf
> taking suggestions from the net (voip-info.org in particular). I changed 
> iptel's position from friend to peer. I turned on and off nat, I chose 
> different codecs in first place, entered my outward IP as fromdomain and 
> uncommented the register directive with correct values.
>    All I get is two registrations now, but no calls.  get a registration
effort
> every 225secs and it succeeds. But when I make a call;
> channel originate sip/iptel-out/echo at iptel.org Application playback 
> vm/net_ring
>    The call is onlyleft in state ACK for a while. Then asterisk tells me,
that
> it is destroying the sip dialog (long ID) INVITE.
>    Question: Might it be a problem, that my system only knows itself as 
> 192.168.*. Do I need to set something else than externip?
>   
Does the server see your sip client at 192.168.*.*? that would be a
problem.
>    Might it be, that my router really blocks certain ports? I can't
check it,
> since it's heavily javascript based and, since I'm blind and the
accessibility
> software for the GUI never really worked on this distro, I don't have a
> browser to look at it.
>   
It's possible that the router is not SIP friendly or there is a setting
to allow sip on it. I can not tell as I don't know what router you are
using.
>    Do I need to forward port 5060 to my machine specifically (like it is
needed
> for SSH's port 22), or is the mechanism based on: I talk first and the
sever
> gets back to me based on that.
>   
Should not need any forwards. However the router could be firewalling
some ports, like the rtp ports. You need to ask what ports are needed
for rtp.

Lyle Giese
LCR Computer Services, Inc.
>    This configuration worked for googletalk. I admit, there were problems,
but
> calls were coming through from both sides.
>    Please can someone help me clear up this mess. I'm completely
frustrated and
> don't know what else to do, where else to look.
>    Kindly yours and thanks in advance
>              JUlien
>
> --------
> Music was my first love and it will be my last (John Miles)
>
> ======== FIND MY WEB-PROJECT AT: =======> http://ltsb.sourceforge.net
> the Linux TextBased Studio guide
> ======= AND MY PERSONAL PAGES AT: ======> http://www.juliencoder.de
>
>

At 01:16 PM 6/5/2010, you wrote:
>    Please can someone help me clear up this mess. I'm completely 
> frustrated and
>don't know what else to do, where else to look.
I've always forwarded port 5060 and all the RTP ports, in my case 
16000-16100, directly to my Asterisk box and I've never had problems. 
For SIP you really need to forward the RTP ports you expect to use 
and you might as well forward 5060 as that's the only place you want it to
go.

Or, that's my belief. Try it and see, but find out what RTP ports you 
need by looking in rtp.conf for the lines:

rtpstart=16000
rtpend=16100

I picked those numbers because I thought using the default huge range 
made no sense and I've never had a problem with my no more than 3 
calls at a time world.

Ira

Hello Ira!
   I will have a look at my rtp.conf and change the rtp-port range there. As to 
forwarding: Well it remains to be seen - pardon the pun - if I can find 
someone willing and patient enough to be my pair of eyes. :-)
   Kindly yours
            Julien

--------
Music was my first love and it will be my last (John Miles)

======== FIND MY WEB-PROJECT AT: =======http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: ======http://www.juliencoder.de

Hello everyone!
   So now I found someone to forward the ports 5060 and 16000-16100 on my 
router and made sure to enter these ports 16000-16100 in rtp.conf. Still I get 
no calls going.
   The call is initiated. "sip show channels" shows the call with
status ACK
and then the dialog with method invite is destroyed.I tried both using 
application jack and application playback.
   So what else can be the problem?
   Kindly yours
           Julien

--------
Music was my first love and it will be my last (John Miles)

======== FIND MY WEB-PROJECT AT: =======http://ltsb.sourceforge.net
the Linux TextBased Studio guide
======= AND MY PERSONAL PAGES AT: ======http://www.juliencoder.de

On Sat, 2010-06-05 at 22:16 +0200, Julien Claassen
wrote:
> But when I make a call;
> channel originate sip/iptel-out/echo at iptel.org Application playback 
> vm/net_ring
>    The call is onlyleft in state ACK for a while. Then asterisk tells me,
that
> it is destroying the sip dialog (long ID) INVITE.
This could be caused by a number of reasons, but the most likely is that
your syntax isn't correct above.  Try either:

channel originate sip/iptel-out/echo Application playback vm/net_ring

or 

channel originate sip/echo at iptel-out Application playback vm/net_ring

--
Jared smith
Digium, Inc.