Help with setup Linksys pap2t, analog phone and *

Hi all,
I am a newbie and am testing and learning my * server. I have a pap2t ata box, a plain ole analog phone (not connected to pstn at all) and my * server running on a static ip. Goal is to talk eventually ip to ip but first just to asterisk. My problem is, i cannot get anything. I am using * bus. ed .1.2.9 running on FC4 OS. Am I supposed to put the mac address and dynamic ip address in sip.conf somewhere, or some other file?.. Or is this just not doable due to the fact that it’s just a dead analog phone? Now, to give a little background, I have already tested * on two sip phones sitting behind two different nat’d firewalls and this test was conclusively successful, as they both registered fine and worked , so now I am moving on to this scenario. Also, my dns server (bind 9.3.1) is already setup and running with the _sip._udp info and is registered with no errors via nslookup and dnsreports, etc.

These are my sip.conf and extensions.conf files:
exten => 1000,1,Dial(SIP/${EXTEN},30)
exten => 1000,2,Hangup


Thanks in advance for any and all help.

try taking this a step at a time.

your asterisk setup works with softphones on your local network right ?
now go to the web interface on the PAP2 and setup 1 user/line to register with your asterisk box. does it register ?
once you have it registering (you can see it happening on the CLI if you have the verbosity set high enough), then try using the analogue phone to a test extension/application.

Thanks for the quick response. What web interface? This was not indicated on the pap2t simplified direction sheet so i went to my router’s web interface but there was no indication of a setup like I thought in this area that I could see. Can you tell me where this web interface is? And, yes, * server setup works great with the sip softphones.

we’re talking about a linksys PAP2T ? unless they’ve reduced the firmware drastically, it’s a net device and you’ll find the web interface at

check the DHCP log of whatever is giving out addresses on your network.

Thank you for your help. I got a dial tone now. However, when I try to call my sip softphones from this phone, all i get after a few moments of waiting if a busy signal. Any ideas?
Thanks in advance.

like i said earlier, one step at a time. try creating an extension that plays a sound file. when you get that working you can try calling another UA.

Thanks, i got it.