I just setup an Asterisk system on a small Shuttle box; I am only using SIP channels and have no FXO/FXS cards. The system works fine in that I can call my inbound number (Broadvoice) and have the system answer and I can make outgoing calls. The problem is that every time I want to change something in the sip.conf file, I have to do a 'restart now' instead of a 'reload' or 'sip reload' as it hangs and stops processing calls or responding on the CLI. I tracked this down to something dealing with the peers I have in sip.conf. However, I removed all peers and just placed a simple friend in sip.conf (right from the sample file):

[xlite1]
;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
regexten=1234 ; When they register, create extension 1234
username=xlite1
callerid="Jane Smith" <5678>
host=dynamic
nat=yes ; X-Lite is behind a NAT router
canreinvite=no ; Typically set to NO if behind NAT
disallow=all
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw



From the Asterisk CLI, I do:

sip show peers

It shows the one peer

and then sip reload

and then

sip show peers

and it is blank with the system being hung.

I've searched high and low for a solution, reinstalled Linux (Fedora Core 3), reinstalled Asterisk (I enabled the ztdummy module), and have used the sample config with the same results (for the sample, I just uncomment the above). Since I used the sample without modifications, the fact that I'm using it behind NAT shouldn't make a difference.


Any ideas on what is causing this? Is there any additional information I can provide for assistance?


Thanks.

--
Scott Gruby
<mailto:[EMAIL PROTECTED]>
<http://www.gruby.com/>

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