[Asterisk-Users] Can Asterisk act like a normal sip phone?

[Walter Doerr]

On Fri, Jan 30, 2004 at 12:06:47PM +0100, Klaus-Peter Junghanns wrote:
> hi,
> 
> just signed up and it works like a charm. :-)
> They even support g711 :) and multiple channels :)
> 
> make sure you have in sip.conf:
> 
> register => XXXX:YYYY at sipgate.de/<extension in your context>
> 

I believe that I have that entry in sip.conf. Maybe not the extension.
Still no luck.


> you will get the too many hops if you try to register
> with their proxy (proxy.de.sipgate.net).

I tried both. Besides, both names resolve to the same IP.


Here is what I just received:

Sip read: 
SIP/2.0 483 Too Many Hops
Via: SIP/2.0/UDP 212.102.234.130:5060;branch=z9hG4bK7ba3fc35
From: "asterisk" <sip:asterisk at 212.102.234.130>;tag=as2f4213d0
To: <sip:217.10.79.9>;tag=b11cb9bb270104b49a99a995b8c68544.566f
Call-ID: 5d101e59279eb0ca0c999ea3337f6299 at 212.102.234.130
CSeq: 102 OPTIONS
Server: Sip EXpress router (0.8.12 (i386/linux))
Content-Length: 0
Warning: 392 217.10.79.9:5060 "Noisy feedback tells:  pid=30305
req_src_ip=217.10.79.9 req_src_port=5060 in_uri=sip:217.10.79.9
out_uri=sip:217.10.79.9 via_cnt==22"


-Walter



-- 
  Walter Doerr   =*=   wd at infodn.rmi.de   =*=   FAX: +49 2421 962001
              "The poor folks who only have 100MBytes of RAM five years
from now may not be able to buffer a 16MB packet, but that's their
tough luck."              (John Gilmore on Mon, 10 Oct 88 18:10:21 PDT)