How to reset linksys pap2t ata

Hi all
I began my setup of the linksys pap2t today. I gained a dial tone but that was it. After dialing an extension, it just rang as busy. Anyway, there was no user guide provided so i went to linksys site and found one but they only had one available for pap2. I followed their instructions ( geared for vonage) and set a static ip (to my * server’s static ip) even though there is no static ip on this particular machine. After doing so, the web interface was lost and line 1 no longer lights up, only the ethernet and power lights up now. I have unplugged power to pap2t, cable dsl box, and router, and rebooted machine several times. I am now just lost. Any help would be appreciated. I do no know how to reset the linksys pap2t now with line 1 no longer available.

Thanks in advance for any assistance.

try to download from linksys manula for pap2
(pap2 and pap2t can’t be so diffrent)
dial **** to get to pap2 IVR
set ip using IVR or do manual (732668) reboot

Hi,
I cannot dial **** because line is no longer lit. I believe the firmware is totally deleted now and I have written linksys. When i go through my router web interface, the pap2t is no longer shown. And when attempting to upgrade firmware, which I had done prior, the firmware window is entirely blank.

I have gone through many tribulations of resetting by unplugging ata, router, dsl box over and over again, but i can no longer get line1/line2 or a web interface, or a mac address or ip address generated by my router. Again, i believe this is because the firmware is no longer available on the ata.

Thanks for your response.

Hi all,

Just in case anyone has a similiar situation and the pap2t box appears compeletely dead to the point that even the firmware appears deleted, it’s not, just dial ****73738# even though there is nothing but dead air. Dialing **** will not work, in and of itself. I must say thanks to fdragowski because you were actually right on target, but i guess the code you suggested only works for pap2.

I found pdf “SPA User Guide (July 2004)” Sipura Technology INC.
ther are list of IVR codes 732668 (reboot) 73738 (reset [factory reset])
almost all codes works for Linksys PAP2. I guessed that thisshouldl work for PAP2t.

Hello all,

I still only get a dial tone and nothing else. My two sip softphones respond with 55000@voipme2u.com cannot be found after trying to call my analog phone connected to the ata. My asterisk server’s console responds with: Jun 25 23:54:19 NOTICE[10437]: app_dial.c:1040 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination) Is it possible to make my analog phone call ip to ip without a pstn line or pri card? Or is ths response actually saying i need to build up on my dialplan or…?

With regard to the ata, the only area I configured in the web interface was line1 with username, password, and in the web interface to the router i only added the udp port information to the ata’s ip address. Is this about all i need to do in this area? I did use the directions for the pap2 to configure this area but… Or do i need to build up on my dialplan more to get additional steps working?

I am experimenting with all of this and reading much, as well as learning much and i greatly appreciate all of the feedback from the forum. I plan to eventually add a DID but am not sure if I am off base with my plans here.

These are my sip.conf and extensions.conf files:
extensions.conf
[general]
static=yes
writeprotect=yes
[internal]
exten => 55000,1,Dial(SIP/${EXTEN},30)
exten => 55000,2,Hangup

sip.conf
[55000]
type=friend
regexten=55000
secret=55000
host=dymamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
dtmfode=rfc2833

pap2 is not registered in asterisk.
there is ‘info’ tab and ‘line 1 status’ section on pap2 www. Registration State: value should be online.
line 1 tab, line enable=yes, sip port=5060, proxy=Your_asterisk_IP, UserID=55000, password=55000
i have linksys pap2, not pap2t but configuration should be very similar

this kind of worries me. your PAP2 has the same IP address as your Asterisk server ??

in your shoes i would reset the PAP2 to factory defaults. assuming it’s DHCP-enabled by default (i can’t remember on mine), make sure DHCP is running on the network. let it get an address. examine the DHCP log or lease listing to see what address it has. then connect to the web interface and follow the wizard at voxilla.com/pap2config.php

Hi,

Here is a portion of my Info tab, which i changed the exten back to 1000, rather than 55000, and in the pertinent * files.
Display Name: 1000 User ID: 1000
Hook State: On Registration State: Online
Last Registration At: 6/26/2006 05:13:11 Next Registration In: 190 s
Message Waiting: No Call Back Active: No
Last Called Number: Last Caller Number:
Mapped SIP Port:
Call 1 State: Idle Call 2 State: Idle
Call 1 Tone: None Call 2 Tone: None

Line1:
Line Enable: yes
SIP Port: 5060
Proxy and Registration
Proxy: voipme2u.com Register: yes
Make Call Without Reg: no Register Expires: 3600
Ans Call Without Reg: no
Subscriber Information
Display Name: 1000 User ID: 1000
Password: ********** Use Auth ID: yes
Auth ID: 1000

My ata has been noticed on the router end of this nat’d firewall but it does not appear to be seen in its entirety on asterisk, but it is registering according to this Info page, so I don’t understand why * is still not acknowledging it – but you’re right about it appearing not to be registered. The asterisk console does not even print out that it received this register. Hum…now what to do?

Oh, and baconbuttie my pap2t does not have the same ip address as my * server. I had not known what to do back then, that now has been resolved - thanks for the follow-up though.

why do you have the proxy set to that ? o that a hostname the PAP2 can resolve and is your Asterisk box ? or am i more confused than ever ?

(it’s perfectly possible … a trip to the hospital last week has left me in lots of pain and with lots of bruises !)

Hi,
voipme2u.com is the hostname/domain name and my * server’s name (resolving name - but actually i tried it using the ip address as the proxy too and i get the same response). I run my own dns server (bind 9.3.1 and allocated _sip._udp to *) and my other sip softphones are also set to it which work fine. I just prefer resolving by name rather than ip’s. I’m not quite sure i understand your questions.

i’m just trying to establish whether your PAP2 is registering to your Asterisk box. your PAP2 would appear to think it is registered, but does Asterisk agree with this ? what does sip show peers return ?

Hi,
This is what sip show peers reported and thanks:
Name/username Host Dyn Nat ACL Port Status
8765/8765 (Unspecified) D 0 Unmonitored ; not finished with this
portsip/portsip (Unspecified) D 0 Unmonitored ; not finished with this
tracy/tracy 68.234.55.244 D 5060 OK (39 ms)
kevin/kevin 24.38.44.74 D 5060 OK (8 ms)
1001 (Unspecified) D N 0 Unmonitored ; not setup yet either
1000/1000 68.234.55.244 D N 15060 Unmonitored
6 sip peers [6 online , 0 offline]

I do not know why it shows a port of 15060 when i put 5060 for the ata unit and my router for the ata’s ip address, hum?

Any ideas…

ok, i think you had better draw us a nice little diagram of how you have your hardware arranged. tracy/tracy is registered from the same address without NAT, but the PAP2 is set to use NAT and isn’t registering ???

Hello,
Let me try and explain, it’s easier for me. Well even though that particular mahcine is beihnd a nat, i have my own dynamic dns updating service that keeps that address appearing static which is why i can allocate nat or not nat (at least for the sip phone, but i am not sure about the ata box). i hope that makes some sense. But what I don’t understand is the 15060 port. This is why the tacy/tracy sip phone works on port 5060 even though its not allocated as nat. I am now wondering if I need to move to another machine becuase that port is already allocated to another sip phone and on the same ip address?

Hi all,

and thanks baconbuttie with your suggestion of "sip show peers’. I switched the pap2t to another machine because the address for the other machine, although the router dishes out ip’s, the machine is still held static and overrides the router. Now I get what i should get, so i can no go to the nest step in learning asterisk. Thanks so much.

Here is the new sip show peers info:
*CLI> – Registered SIP ‘1000’ at 69.160.58.221 port 5060 expires 3600

*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
8765/8765 (Unspecified) D 0 Unmonitored
portsip/portsip (Unspecified) D 0 Unmonitored
tracy/tracy 68.234.55.244 D 5060 OK (27 ms)
kevin/kevin 24.38.44.74 D 5060 OK (14 ms)
1001 (Unspecified) D 0 Unmonitored
1000/1000 69.160.58.221 D N 5060 Unmonitored
6 sip peers [6 online , 0 offline]

but the unmonitored looks suspicious, but is this the normal output when behind a nat’d firewall?

Asterisk will only monitor a peer if you use “qualify=yes” (or a timeout). there’s usually no need for a UA.

hi all,

thanks again. I still only get a dial tone. I can not dial my other two sip softphones using this analog/ata phone nor can my other two sip soft phones dial it, they both say: 1000@voipme2u.com not found, athough it is now officially registered with my * server. I will continue to research but if you know of any reason, please let me know. The simple dial plan (although not so simple to me) i have is listed below:

[general]
static=yes
writeprotect=yes

[internal]
;exten => 1000,1,Dial(SIP/${EXTEN},30) ; dial extension so you dont have to
;exten => 1000,1,SIPAddHeader(“Alert-Info: http://voipme2u.com/sound.wav”)
exten => 1000,1,Dial(SIP/1000, 30)
exten => 1000,2,Hangup ; 30 is how long (30 sec) it will dial before going
; to next priority (2) which disconnects the call.
exten => 1001,1,Dial(SIP/1001,30)
exten => 1001,2,Hangup

sip.conf
[1000]
type=friend
regexten=1000 ;username
secret=************
host=dynamic
nat=yes
port=5060
mailbox=1000
qualify=yes
canreinvite=no
disallow=all ; recommend ulaw/alow for now (g711)
allow=ulaw
allow=alaw
dtmfmode=rfc2833
callerid= ‘Linksys’ <1000> ; caller id name and number/extension
context=internal ; reference internal context in extensions.conf

there’s no “context=” for your entries in sip.conf. is this deliberate ?

Hi,
I have context=internal to reference it in extensions.conf. Oh i see it was not listed in the first post - a bad copy and paste on my part – apologies.