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mberlant Dan Should Pay Me

Joined: 01 Feb 2009 Posts: 793 Location: Japan
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Posted: Fri Mar 06, 2009 7:05 pm Post subject: |
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They may be elsewhere on a PAP2 (I don't have one nearby), but on all SPA devices these are fill-in boxes at the top of the User page.
The first section of boxes lets you fill in the destination telephone numbers for CFA, CFB and CFNA and the ring delay for CFNA. The second section of boxes lets you fill in a Caller ID number or pattern in the left-hand box and the destination phone number for that pattern in the right-hand box.
These functions of the ATA work by intercepting the Incoming Call packet and sending it back to the SIP server as a Redirect packet containing the new destination telephone number.
Since the MJ softphone does not work this way (Call Forwarding is performed by registering the request inside MJ's server), causing your ATA to send this illegal packet to the MJ server is an invitation to be identified as a fraudulent user.
There is another reason I would not like to see you tempt the fates by trying this feature. One of the ways we get to continue using our own ATAs with the MJ service is by staying under MJ's radar. I don't recommend anything that has a high probability of waking that sleeping bear. |
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Posted: Fri Mar 06, 2009 7:05 pm Post subject: Magicjack support, tips, tricks, and hacks |
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MagicDump Dan isn't smart enough to hire me
Joined: 11 Sep 2008 Posts: 100
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Posted: Fri Mar 06, 2009 7:22 pm Post subject: |
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Thanks mberlant:
I have edited the first post and posted the screen shot of the User Menu and as you said is in the first part of the User Menu as Cfwd options.
So just don't put anything in there. |
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MagicDump Dan isn't smart enough to hire me
Joined: 11 Sep 2008 Posts: 100
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Posted: Fri Mar 06, 2009 7:55 pm Post subject: |
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| pagemen wrote: | the correct daylight saving rule should be
start=3/8/7/2:00;end=11/1/7/2:00;save=1 |
Thanks pegemen, the correction have been made. |
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FrankPepe magicJack Apprentice
Joined: 15 Feb 2009 Posts: 13
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Posted: Sat Mar 07, 2009 1:47 am Post subject: |
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Hi Msiam
, My RTP300 was working great untill 3 days ago and then I started experiencing this intermittent one way audio you describe. have you found a solution in the settings yet? If not in the settng is ther a thread devoted to this topic elsewhere? thanks!
-Frank
| 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 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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mberlant Dan Should Pay Me

Joined: 01 Feb 2009 Posts: 793 Location: Japan
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Posted: Sat Mar 07, 2009 1:58 am Post subject: |
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| Yes, someone in another thread said that he had success by changing which proxy server he pointed to. Search for "proxy" and scan the titles for that recent posting if you want to read that discussion. |
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AlpineMan magicJack Apprentice
Joined: 02 Mar 2008 Posts: 12
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Posted: Sat Mar 07, 2009 1:32 pm Post subject: |
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| 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!) |
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Does this dial plan require the caller to enter area code first even if the number they're calling is in the same area code? Example, my area code is 626...and I want to call a friend in the 626 area code. Will I need to dial 1-626-xxx-xxxx or can I just dial xxx-xxxx using this dial plan. Thanks! |
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msiam Dan isn't smart enough to hire me

Joined: 15 Nov 2007 Posts: 476 Location: WI
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Posted: Sat Mar 07, 2009 1:51 pm Post subject: |
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Frank, It seems that my problems have disappeared, so far, ever since I changed the proxy location, I will monitor and if I run into more difficulties, I will try to change the proxy again. So, it's " So Far, So Good", although, It just could have just been an upgrade issue from MJ for the last week or two.. Anyway.. we'll see. _________________
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mberlant Dan Should Pay Me

Joined: 01 Feb 2009 Posts: 793 Location: Japan
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Posted: Sat Mar 07, 2009 6:53 pm Post subject: |
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Discussion in another thread caused me to notice that there is no STUN setting in this configuration. While the ATA has other tricks to discover its public information in order to give that information to the SIP server, STUN is a very reliable method when these tricks fail for one reason or another.
I recommend the following settings at the bottom of the SIP page:
STUN Enable: Yes
STUN Test Enable: No
STUN Server: stun.ekiga.net:3478
NAT Keep Alive Intvl: 15
If this helps anyone close up the security exposure of forwarded ports, that would make me very happy. |
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mberlant Dan Should Pay Me

Joined: 01 Feb 2009 Posts: 793 Location: Japan
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Posted: Sat Mar 07, 2009 6:55 pm Post subject: |
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| AlpineMan wrote: | | Does this dial plan require the caller to enter area code first even if the number they're calling is in the same area code? Example, my area code is 626...and I want to call a friend in the 626 area code. Will I need to dial 1-626-xxx-xxxx or can I just dial xxx-xxxx using this dial plan. Thanks! | Replace 311 in the third entry of the Dial Plan with 626. I explained this in the original thread, but didn't carry the whole discussion here. |
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MagicDump Dan isn't smart enough to hire me
Joined: 11 Sep 2008 Posts: 100
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Posted: Mon Mar 09, 2009 8:24 am Post subject: |
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Public STUN servers
* stun.ekiga.net
* stun.fwdnet.net (no XOR_MAPPED_ADDRESS support)
* stun.ideasip.com (no XOR_MAPPED_ADDRESS support)
* stun01.sipphone.com (no DNS SRV record)
* stun.softjoys.com (no DNS SRV record) (no XOR_MAPPED_ADDRESS support)
* stun.voipbuster.com (no DNS SRV record) (no XOR_MAPPED_ADDRESS support)
* stun.voxgratia.org (no DNS SRV record) (no XOR_MAPPED_ADDRESS support)
* stun.xten.com
* stunserver.org see their usage policy
* stun.sipgate.net:10000
* numb.viagenie.ca (http://numb.viagenie.ca) (XOR_MAPPED_ADDRESS only with rfc3489bis magic number in transaction ID)
* stun.ipshka.com inside UA-IX zone russsian explanation at http://www.ipshka.com/main/help/hlp_stun.php |
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VoipDude Dan isn't smart enough to hire me

Joined: 06 Mar 2009 Posts: 129
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Posted: Wed Mar 11, 2009 7:47 pm Post subject: |
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Problem configuring my ATA. It's probably something simple I am overlooking. I'll try again later.
Last edited by VoipDude on Wed Mar 11, 2009 9:09 pm; edited 1 time in total |
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VoipDude Dan isn't smart enough to hire me

Joined: 06 Mar 2009 Posts: 129
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Posted: Wed Mar 11, 2009 8:03 pm Post subject: |
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I must say you surely have to have patience when your working with this kind of stuff 
Last edited by VoipDude on Wed Mar 11, 2009 9:11 pm; edited 2 times in total |
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pagemen Dan isn't smart enough to hire me
Joined: 15 Dec 2008 Posts: 128
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Posted: Wed Mar 11, 2009 8:57 pm Post subject: |
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It would be great if someone could upload the configuration file backupped by this utility
http://www.opensky.ca/~jdhildeb/software/spaconf/
just make sure to remove the phone#.
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you can back up your configuration to a computer
you can more easily swap configurations with other users
you can compare configurations easily (using diff)
you can use this tool to update your configuration programmatically (for different times of day, etc.)
you can store your configuration in a source control system
you prefer editing text files to using web interfaces
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VoipDude Dan isn't smart enough to hire me

Joined: 06 Mar 2009 Posts: 129
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Posted: Wed Mar 11, 2009 9:03 pm Post subject: |
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Problem solved.
Thanks to MagicDump for his excellent configuration picture post.
Thanks to mberlant for encouraging me to not give up.
VoipDude |
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MagicDump Dan isn't smart enough to hire me
Joined: 11 Sep 2008 Posts: 100
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Posted: Wed Mar 11, 2009 11:37 pm Post subject: |
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| pagemen wrote: | | It would be great if someone could upload the configuration file backupped by this utility |
Thanks pagemen, excellent program, here is a backup file:
http://rapidshare.com/files/208210096/PAP2TMJ.config
You can open it as a text file.
Just ignore line2, and replace your SIP Credentials.
Enjoy |
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