configuring linksys pap2 w/ SIP info

bazilla901 · Post by **bazilla901** » Sat Apr 12, 2008 12:39 am

Having problems setting it up to work, i get dial tone but any number i dial i get busy signal, what do i put in registrar address and port?

SIP tab

Proxy Server : Port:
Registrar Address : Registrar Port:
Notify Interval : ms

These are the main boxes on Line 1

Caller ID Number: Caller ID Name:
Proxy and Registration
User Name: Password:
Proxy Server: Proxy Port:
Registrar Address: Registrar Port:
Dial Plan
Digit Map:
Termination Digit: Partial Digit Timer: sec
Critical Digit Timer: sec Dial Timeout: sec
Preferred CODEC
Voice Coding Profile : Fax Relay Profile:
Modem Relay Profile: P-Time Voice :
P-Time Data:

gooney · Post by **gooney** » Sat Apr 12, 2008 12:48 am

bazilla901 wrote:Having problems setting it up to work, i get dial tone but any number i dial i get busy signal, what do i put in registrar address and port?

SIP tab

Proxy Server : Port:
Registrar Address : Registrar Port:
Notify Interval : ms

These are the main boxes on Line 1

Caller ID Number: Caller ID Name:
Proxy and Registration
User Name: Password:
Proxy Server: Proxy Port:
Registrar Address: Registrar Port:
Dial Plan
Digit Map:
Termination Digit: Partial Digit Timer: sec
Critical Digit Timer: sec Dial Timeout: sec
Preferred CODEC
Voice Coding Profile : Fax Relay Profile:
Modem Relay Profile: P-Time Voice :
P-Time Data:

Search the forums and you will find the answer.

bazilla901 · Post by **bazilla901** » Sat Apr 12, 2008 1:00 am

I have read all the posts related to setting up an ATA and it is not helping. I couldn't find what to put in "registrar" if you could pelase refer me to a thread or explain ifyou can thanks.

Stewart · Post by **Stewart** » Sat Apr 12, 2008 2:15 am

You have a PAP2 v2, which has settings very different from the classic PAP2.

Set Registrar Address same as your Proxy Server, e.g. proxy1.nashville.talk4free.com

Set Registrar Port and Proxy Port to 5070

Caller ID Number and User Name to e.g. E212867530901
Password to your 20-character SIP password

If it still doesn't work, report whatever status you can get out of the device. If it has a SIP Debug feature, that may be useful, or you may need to capture traffic to see what is going wrong.

bazilla901 · Post by **bazilla901** » Mon Apr 14, 2008 3:38 pm

hrmm still not working

enotz · Post by **enotz** » Wed Apr 30, 2008 3:23 pm

I have my credentials, I know my pap2T works well since line2 is operational and works well with sipphone.

What do I need as my settings for line1 to use my MagicJack account.

I have my:
ProxyUserName=E303449914901
ProxyUserPassword=BXK1Z468228WWCV7CNAG
SIPCallerID=E303449914901
RegisterOnProxy=1

I used the E... as the userID and the BXK... as the password, set to port 5070 and hoped. But it tells me registration state: offline.

BTW, I did change a couple of numbers above, for obvious reasons

LikeMagic · Post by **LikeMagic** » Wed Apr 30, 2008 3:26 pm

I hope the above isn't your real MJ SIP credential

enotz · Post by **enotz** » Wed Apr 30, 2008 3:27 pm

LikeMagic wrote:I hope the above isn't your real MJ SIP credential

I changed a couple of numbers for obvious reasons, but thanks for the heads up, I might have been an idiot

LikeMagic · Post by **LikeMagic** » Wed Apr 30, 2008 3:41 pm

How do you enter the proxy server address and proxy port?

Are they on separate lines?

Or in the same line separated by ":" ? eg: proxy1.nashville.talk4free.com:5070

enotz · Post by **enotz** » Wed Apr 30, 2008 4:39 pm

I had not put the :5070 on the end of the proxy1.... - I had put it there as the sip port and that's what confused me - so thanks.

onlinepcfun · Post by **onlinepcfun** » Wed May 14, 2008 10:27 am

Hi Stewart...I am having the same issue with PAP2 v2... looks like PAP2 v2 does not like alpha charecters in UserID...it seems to be replacing with * - do you know any way to escape that replacement it is doing...

Any idea about user agent string that needs to be passed to MJ...I have unlocked Vonage device with downgrading to 1.00.13 and used CYT to unlock it and seems it changed/appended something to PAP2...

I have firmware 1.00.13..not sure if it's the issue with that firmware...

Here is my other thread on this issue:
http://www.phoneservicesupport.com/magi ... t1528.html

Stewart wrote:You have a PAP2 v2, which has settings very different from the classic PAP2.

Set Registrar Address same as your Proxy Server, e.g. proxy1.nashville.talk4free.com

Set Registrar Port and Proxy Port to 5070

Caller ID Number and User Name to e.g. E212867530901
Password to your 20-character SIP password

If it still doesn't work, report whatever status you can get out of the device. If it has a SIP Debug feature, that may be useful, or you may need to capture traffic to see what is going wrong.

Stewart · Post by **Stewart** » Wed May 14, 2008 10:42 am

I came across this: http://www.bellshare.com/en/support.php?folder=17 .
Try setting caller ID both empty and to your Exxxxxxxxxx01 value.

If no luck, you might try newer firmware (at risk of relocking, perhaps permanently).

Or, one could rig a script to fix up the SIP, but that would require leaving a computer on all the time.

You could put the fixup into an open-source router, but that would take a lot of effort.

Sorry, I don't have any good ideas on this.

onlinepcfun · Post by **onlinepcfun** » Thu May 15, 2008 10:38 pm

Thanks Stewart for the suggestion...was able to go with RTP300 firmware on this and solve this issue...
Instructions from from http://www.dslreports.com/forum/r19868031-

Now it registers with MJ perfectly...

Stewart wrote:I came across this: http://www.bellshare.com/en/support.php?folder=17 .
Try setting caller ID both empty and to your Exxxxxxxxxx01 value.

If no luck, you might try newer firmware (at risk of relocking, perhaps permanently).

Or, one could rig a script to fix up the SIP, but that would require leaving a computer on all the time.

You could put the fixup into an open-source router, but that would take a lot of effort.

Sorry, I don't have any good ideas on this.

somenon3 · Post by **somenon3** » Mon May 19, 2008 8:49 am

Hello all,
I guess I could be considered a "newbie" at magicJack and using an ATA. Have used Vonage for years, but I couldn't pass up the MJ pricing.
I've been researching this forum for a while, and it has helped me with discovering my SIP credentials (which I will not post, from what I've seen MJ monitors this forum, and posting the process would just screw the guys that have figured it out). On that note, guys from MagicJack: You guys need to go ahead and let everyone use BYOD. Your methods will continue to be discovered, and is it really worth your time to prevent users from doing this, as it'll probably bring more revenue from those of use who want to use an ATA, and reduce your cost of re-writing that software every time someone figures it out? Yes, it would reduce your forced advertising revenue, but it would increase savings by reducing hardware & tech. support costs.
Anyway,
I have been able to configure my RT31P2 (unlocked) with MJ. I have been able to register the device and make outbound calls on it as well.
Everything works great, sound quality is good, but I have one issue, and I don't know if anyone can help me on this one:
I can't get incoming calls to work. I get this message that states "you have dialed an invalid number (or something like that), announcement 14, switch 179-1". It doesn't sound like an MJ recording (or at least none that I've heard so far).
I know I'm getting at least some SIP packets into my box, as the "bytes received" count increases by about 1K (or so) when I make an inbound call, but then it just quits communicating. My guess is that the RT31P2 box is having issues "handshaking" with MJ's servers. I have the "NAT Keep Alive" turned on, and since I am getting some packets in, I believe it's not a NAT/Firewall issue.
Anybody got an idea of what to check. I have attached screenshots of the SIP tab, and the Line2 and User2 (I'm using Line2) settings:

onlinepcfun · Post by **onlinepcfun** » Mon May 19, 2008 10:37 am

Here are the settings I have used on PAP2v2 with RTP300 firmware on it

From Default settings - NOT CHANGED ANY EXCEPT the following so please revert any settings that you might have done to test this.
Proxy: YOUR MJ PROXY
Use Outbound Proxy: YES
Outbound Proxy: same as above proxy
Display Name: Anything you like (Doesn't get used for PSTN calls - I believe)
UserID: EXXXXXXXXXX01 (MJ's user ID)
Password: UR MJ password
Use Auth ID: YES
AuthID: Same as above user ID

Stewart · Post by **Stewart** » Mon May 19, 2008 11:11 am

somenon3 wrote:I know I'm getting at least some SIP packets into my box, as the "bytes received" count increases by about 1K (or so) when I make an inbound call, but then it just quits communicating. My guess is that the RT31P2 box is having issues "handshaking" with MJ's servers.

Don't work in the dark. Set SIP Debug Option to full, set Debug Server to the address of your PC. Capture with Wireshark. If you tell Wireshark to decode UDP destination port 514 (normally syslog) as SIP, the packets should get correctly parsed.

If you want to take some quick potshots:
Turn off NAT keep alive, but set Register Expires to 50.
Set SIP Port to unique values for Lines 1 and 2, e.g. 5060 and 5061, respectively.
Set Proxy to magicjack.com , Outbound Proxy to proxy1.nashville.talk4free.com:5070 , Use Outbound Proxy to yes.

If no luck, in addition to Wireshark results, post what kind of router and modem you have.

somenon3 · Post by **somenon3** » Mon May 19, 2008 11:17 am

Thanks guys, I'll have to post it later tonight, I VPN'd into my home router from work to get those screenshots, but I'll have to run wireshark on my home pc (which VNC'ing to it won't work well for this situation).
I don't know if you guys caught this or not, but i just wanted to reitterate: I can make outbound calls with no problems at all, I get some SIP packets on inbound calls, but all I get in response is "you have reached a non-working number, announement 14, switch 179-1".

I'll try the suggestions from you guys, and report back the results tonight.

Thanks,
-Some

somenon3 · Post by **somenon3** » Mon May 19, 2008 12:29 pm

well, while doing some quick testing from work, I made the following changes, and it seemed to work (although I'm not at home to see if the phone actually rings, BUT, I did get the caller ID and the RINGING state on Line 2)
Use outbound proxy: yes
outbound proxy: same as proxy
changed line 1 sip port to 5060
changed line 2 sip port to 5061
That was it! I'll check it tonight and report my findings

Thanks a million guys!

somenon3 · Post by **somenon3** » Mon May 19, 2008 8:08 pm

It works! Thanks guys, not sure what fixed it (outbound proxy, or separating the lines by port, but it worked)
Call waiting works too.

Thanks again everybody,
-Some

somenon3 · Post by **somenon3** » Mon May 19, 2008 10:32 pm

ok, one final thing:
When I dial my own number to access voicemail, the auto-attendant prompt doesn't sound like it did before, it says "the person at extension XXX-XXX-XXXX-XX is not available" (same message you get when you call someone who's MJ is not online, and I can't press the * key, (I know I can send *, I've tested it on my phone system at work). Anyway, I'm willing to bet it has something to do with something that MJ is checking when the user calls in (maybe the display name?). Was just wondering if anybody had any insight on this.

Thanks,
-Some

Stewart · Post by **Stewart** » Tue May 20, 2008 5:05 am

somenon3 wrote:ok, one final thing:
When I dial my own number to access voicemail, the auto-attendant prompt doesn't sound like it did before, it says "the person at extension XXX-XXX-XXXX-XX is not available" (same message you get when you call someone who's MJ is not online, and I can't press the * key, (I know I can send *, I've tested it on my phone system at work). Anyway, I'm willing to bet it has something to do with something that MJ is checking when the user calls in (maybe the display name?). Was just wondering if anybody had any insight on this.

Thanks,
-Some

Does DTMF Tx Method: Auto help? Does it work from the softphone? When calling your VM from another service?

somenon3 · Post by **somenon3** » Tue May 20, 2008 6:11 pm

that worked, thanks much Stewart, Dan really does need to hire you!

Lakewell · Post by **Lakewell** » Fri Dec 05, 2008 8:18 pm

Don't even know if the posters are around anymore, but this thread helped a lot after I unlocked my RT31P2. Thanks a lot.

magicJack and magicJack Plus Support, Reviews, FAQs and Tricks

configuring linksys pap2 w/ SIP info

configuring linksys pap2 w/ SIP info

Re: configuring linksys pap2 w/ SIP info

making it work with pap2

it was the port number

PAP2 v2

RT31P2 (unlocked)

Re: RT31P2 (unlocked)