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Magicjack in a PAP2T Configuration Pics
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MagicDump
Dan isn't smart enough to hire me


Joined: 11 Sep 2008
Posts: 100

PostPosted: Wed Mar 04, 2009 10:29 pm    Post subject: Magicjack in a PAP2T Configuration Pics Reply with quote












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10 Digits Linksys PAP2T USA and Canada Dial Plan no Long Distance:

Code:
([49]11S0|[2-9]xx[2-9]xxxxxxS0|011!)


Code:
[49]11S0
[49]: Anything enclosed within '[]' brackets represents 1 number. In the above case, it's a number range allowing either a 4 or 9 to fit the dial plan.
In other words, You can dial 411 or 911

S0: (S followed by the number 0) represents 'Straight Out'. So this part of the dial plan is saying to your PAP2 that should a person dial a sequence of keys that 'fit' the above portion of the dial plan, process the call immediately (i.e., without waiting for more digits to be pressed on the keypad).


Code:
|[2-9]xx[2-9]xxxxxxS0|
This part of the Dial Plan will allow yo to dial your 10 digits number without waiting for more digits to be pressed on the keypad.


Code:
|011!
This part of the Dial Plan will prohibit any long distance calling.


(: The entire dial plan must be enclosed within a pair of brackets '()'.

|: The '|' in a dial plan separates each component of that dial plan.


This next calling Plan will avoid any accidental calling to 911.

Code:
(411S0|911!|[2-9]xx[2-9]xxxxxxS0|011!)


!: The '!' at the end of the number will prevent the number for been dialed.

Magicjack in a PAP2T Configuration Text Version.

http://rapidshare.com/files/208210096/PAP2TMJ.config

The above Link is a text version configuration for the PAP2T with Magicjack, just download the file and open it up as a text, with Notepad or any text editor, cut and paste as you need and change your Dial Plan and Sip info accordingly.There is also a program mentioned in this Post (Spaconf) that will allow you to backup and also restore your PAP2T configuration using this or any or your own file fast an easy, but that will be another Post discussion.

You can download Spaconf from this following link:

http://www.opensky.ca/~jdhildeb/software/spaconf/downloads/


Last edited by MagicDump on Wed May 13, 2009 6:06 pm; edited 9 times in total
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PostPosted: Wed Mar 04, 2009 10:29 pm    Post subject: Magicjack support, tips, tricks, and hacks


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mberlant
Dan Should Pay Me


Joined: 01 Feb 2009
Posts: 793
Location: Japan

PostPosted: Thu Mar 05, 2009 5:35 am    Post subject: Reply with quote

Everything in this configuration looks wonderful except the Dial Plan. The Dial Plan, from the parts that I can see, will not work as desired and will not prevent calls to 1-900 services because of the ambiguities introduced in the first clause. This code, which I published in a thread dedicated to Linksys/Sipura Dial Plans, has no ambiguities in it, and will prevent calls to 900 and 976 services.
Code:
([1235-9]11!|411S0|<:1311>[2-9]xxxxxxS0|1[2-9]xx[2-9]xxxxxxS0| 1441!|1473!|1649!|1758!|1767!|1784!|1876!|1[26][68]4!|1[28]68!| 124[26]!|134[05]!|167[01]!|18[06]9!|011!)
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MagicDump
Dan isn't smart enough to hire me


Joined: 11 Sep 2008
Posts: 100

PostPosted: Thu Mar 05, 2009 9:39 am    Post subject: Reply with quote

mberlant wrote:
Everything in this configuration looks wonderful except the Dial Plan. The Dial Plan, from the parts that I can see, will not work as desired and will not prevent calls to 1-900 services because of the ambiguities introduced in the first clause. This code, which I published in a thread dedicated to Linksys/Sipura Dial Plans, has no ambiguities in it, and will prevent calls to 900 and 976 services.
Code:
([1235-9]11!|411S0|<:1311>[2-9]xxxxxxS0|1[2-9]xx[2-9]xxxxxxS0| 1441!|1473!|1649!|1758!|1767!|1784!|1876!|1[26][68]4!|1[28]68!| 124[26]!|134[05]!|167[01]!|18[06]9!|011!)


Thank you mberlan

I think Dan should pay you LOL Laughing
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msiam
Dan isn't smart enough to hire me


Joined: 15 Nov 2007
Posts: 476
Location: WI

PostPosted: Thu Mar 05, 2009 11:55 am    Post subject: Reply with quote

mberlantm, what would cause a lot of calls to come in on the PAP2 NA or the InnoMedia (SR) to have a one way audio?? I call the MJ number from the Cell, I hear me on the cell but cant hear the cell.. Confused then I call back from the other and the audio is fine.. Confused
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Last edited by msiam on Thu Mar 05, 2009 12:15 pm; edited 1 time in total
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freebie916
MagicJack Newbie


Joined: 21 Feb 2009
Posts: 7

PostPosted: Thu Mar 05, 2009 12:12 pm    Post subject: User 1 Reply with quote

Did you have to configure anything in the User 1 option menu?
I matched kumar's config as much as possible but this one
was more concise because it was for my model ATA.

I matched all of the settings on this one (I included my account info of course) and I get the same results, no incoming calls. They all
go straight to voice mail. I've been going over all the posts
in this forum. I've set my reg exp from 840 to 60 and still no avail.
I also put the PAP2t into DMZ and still no incoming calls.
Any help would be appreciated. thanks.
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msiam
Dan isn't smart enough to hire me


Joined: 15 Nov 2007
Posts: 476
Location: WI

PostPosted: Thu Mar 05, 2009 12:19 pm    Post subject: Reply with quote

Magicdump, I might suggest you disable the Provisioning , choose "no" In all the other posts on this it is one of the first things to perform.. Correct me if I am wrong, mberlant Confused Also, I question in the "line 1" sip port, shouldn't that be "5060"? The 5070 is entered in the proxy1, Just asking here, I have mine set at 5060 and it is working fine with that, except for that question about the on and off one way audio that I am suspecting is a universal MJ on ATA problem. mberlant, .Please correct me on that if I am wrong..
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MagicDump
Dan isn't smart enough to hire me


Joined: 11 Sep 2008
Posts: 100

PostPosted: Thu Mar 05, 2009 1:01 pm    Post subject: Reply with quote

msiam wrote:
Magicdump, I might suggest you disable the Provisioning , choose "no" In all the other posts on this it is one of the first things to perform.. Correct me if I am wrong, mberlant Confused Also, I question in the "line 1" sip port, shouldn't that be "5060"? The 5070 is entered in the proxy1, Just asking here, I have mine set at 5060 and it is working fine with that, except for that question about the on and off one way audio that I am suspecting is a universal MJ on ATA problem. mberlant, .Please correct me on that if I am wrong..

This is getting interesting I know I will probably be able to improve my settings, but my Pap2t is working very good, no problems with audio now after mberlan point out in one of the threads of the use of just G711u audio codec. I will probably change my Provisioning to "no" but as far as the port I will keep it as 5070.
May be mberlan will like to advise better. Wink
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Poo619
MagicJack Expert


Joined: 20 Nov 2007
Posts: 96

PostPosted: Thu Mar 05, 2009 6:01 pm    Post subject: Reply with quote

I'm using a SPA-2102 and I have both in and out audio. My SIP port is set to 5060. My proxy is set to xxx.xxx.xxx.xxx:5070 (xxx is the IP address for my proxy followed by 5070 for the port.) If you are having 1 way audio please check the Codecs you are using along with your router settigns. Be sure to forward ports 5060-5070 to your sip device.
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mberlant
Dan Should Pay Me


Joined: 01 Feb 2009
Posts: 793
Location: Japan

PostPosted: Thu Mar 05, 2009 6:23 pm    Post subject: Reply with quote

Poo619 wrote:
I'm using a SPA-2102 and I have both in and out audio. My SIP port is set to 5060. My proxy is set to xxx.xxx.xxx.xxx:5070 (xxx is the IP address for my proxy followed by 5070 for the port.) If you are having 1 way audio please check the Codecs you are using along with your router settigns. Be sure to forward ports 5060-5070 to your sip device.
Many people here are confusing the SIP server's "hailing" port with the client device's "listening" port. MJ has defined that all of their proxy servers listen on Port 5070, so that is what you must hail them on, as in "xxx.xxx.xxx.xxx:5070" for the appropriate proxy field. Your own port, "Line1|SIP Settings|SIP Port:" in this example, may be anything you wish, as long as each of the Lines (and features, like web access) within the same ATA is assigned a different port number. It is only convention that says that Line 1 is normally assigned 5060, Line 2 is normally assigned 5061, web access is normally assigned 80, etc. No matter whether you choose 5060, 5070 or 12345 as your own listening port number, your router in doing its job will reassign that transaction to any available Port number as it does its NAT function to send the call out into the public internet.

I am also a firm believer that it is never a good idea to open inbound ports in a router in support of any SIP client, and is only a necessary evil when your router is the second NAT router sitting behind another NAT router (your ISP's or your hotel's or your apartment building's) that you cannot control. There are other threads here that discuss the intricacies of that topic.

By the way, for the purposes of this thread's topic, all of this discussion is applicable to any Linksys or Sipura ATA or SIP phone. As far as one-way audio problems, I recommend addressing those problems in a thread already devoted to that topic, and only bringing the solution back to this thread if it turns out to be a parameter setting in a Linksys/Sipura ATA.
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freebie916
MagicJack Newbie


Joined: 21 Feb 2009
Posts: 7

PostPosted: Thu Mar 05, 2009 9:09 pm    Post subject: Reply with quote

So are we to assume that nothing is changed in the User 1 menu?
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mberlant
Dan Should Pay Me


Joined: 01 Feb 2009
Posts: 793
Location: Japan

PostPosted: Thu Mar 05, 2009 9:50 pm    Post subject: Reply with quote

I can only recommend that you do not program any of the automated Call Forwarding features, because MJ's servers will not honor a REDIRECT message and if you send them one it is a good way to flag your account for fraud and have it banned.

What feature on the User page are you thinking about?
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freebie916
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Joined: 21 Feb 2009
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PostPosted: Fri Mar 06, 2009 2:55 am    Post subject: Reply with quote

Thanks for your help guys! I finally got it to work correctly. I am able to make calls as well as receive them. The settings I had to change were in the User 1 menu.

I had to change the default settings in the User 1 menu's Supplementary Service Settings. I matched them up to Magicdump's settings and it worked perfectly! Thanks again!
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MagicDump
Dan isn't smart enough to hire me


Joined: 11 Sep 2008
Posts: 100

PostPosted: Fri Mar 06, 2009 7:45 am    Post subject: Reply with quote

freebie916 wrote:
Thanks for your help guys! I finally got it to work correctly. I am able to make calls as well as receive them. The settings I had to change were in the User 1 menu.

I had to change the default settings in the User 1 menu's Supplementary Service Settings. I matched them up to Magicdump's settings and it worked perfectly! Thanks again!


freebie916

The settings posted in here for the User1 are the default settings. I didn't post them before because I never change anything in it, but them I realized someone may have changed something accidentally.

I glad I could help Smile
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freebie916
MagicJack Newbie


Joined: 21 Feb 2009
Posts: 7

PostPosted: Fri Mar 06, 2009 5:55 pm    Post subject: Reply with quote

mberlant wrote:
I can only recommend that you do not program any of the automated Call Forwarding features, because MJ's servers will not honor a REDIRECT message and if you send them one it is a good way to flag your account for fraud and have it banned.


I'm a little worried about this one now. Are these call forwarding features on by default? Also what menu are they located on in the PAP2T configs?
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pagemen
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Joined: 15 Dec 2008
Posts: 128

PostPosted: Fri Mar 06, 2009 6:55 pm    Post subject: Reply with quote

the correct daylight saving rule should be

start=3/8/7/2:00;end=11/1/7/2:00;save=1
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