[Asterisk-Users] Dialing SIP

[Benjamin J. Bawkon]

Im really starting to get the hang of Asterisk, however, I still have
one issue...

My SIP Client can dial other extensions just fine, but no extension can
ring the Sip client...

Here is the pertinent info:
SIP.CONF,
[general]
port = 5060
bindaddr = 192.168.0.5 	;ip of asterisk server
context = default

[301]
username=301
context=local
type=friend
secret=test
insecure=yes
host=dynamic

----------------------------------------
EXTENSIONS.CONF

[local]
exten => _1XX,1,Dial,ZAP/1/BYEXTENSION
exten => 301,1,Dial,SIP/sip:301 at 192.168.0.5	; again, ip of * server

blah blah blah below this..

----------------------------------------
Console Debug:

When 301 is Dialed:

--Executing Dial("OSS/dsp", "SIP/sip:301 at 192.168.0.5") in new stack
Called sip:301 at 192.168.0.5
Got SIP response 482 "Loop Detected" back from 192.168.0.5
No one is available to answer at this time
WARNING[114703]: File pbx.c, Line 1268 (ast_pbx_run): Timeout, but no
rule 't' in context 'local'

----------------------------------------
Problem is, the SIP Client never rang.....

Now...If I change the extensions.conf to read:
Exten => 301,1,Dial,SIP/sip:301 at 192.168.0.109

Then it works fine...problem is, 192.168.0.109 is a DHCP'd ip address to
the sip client machine...It will change occasionally...

Any Ideas?  Thanks!
Ben Bawkon