magicJack and magicJack Plus Support, Reviews, FAQs and Tricks

I am glad to hear of your happy ending. Thanks for reporting back.

Hi, does anyone else experience an echo? Can I do anything to reduce the echo? I have high speed internet.

Download 9.52 Mb/s
Upload 2.09 Mb/s
Ping 11 ms

I just got my PAP2T-NA, and am trying to configure it. I followed the instructions on this thread, but I don’t know where do I get the Subscriber Info in line 1 from. (Display Name: Exxxxx01, Password, Auth ID, and User ID).

Do I get this info from my MJ? If yes, how?

Thanks for your help everyone.

1. are you saying if you have constant call forward to a different phone number you can't use the pap2t my magic jack permanently forwards to another number.

2. Is this more reliable that using there dongle I have autodialers on and notice alot of no dial tones and the dongle needs to be reset not the most reliable.


craig

Zen, do a search for Stroths. And read the thread on sip grabber. You can do it but it may take some time.

Thanks for the info Oldtimercurt,

I have tried v1.5 and v1.6 and can't get them to work for me. Here is what I get when I press these buttons:

v1.5
Get Current SIP Info: InvalisMJFlashDriveLetter in Config.ini or MJ not plugged in. Correct and try again
Upgrade MJ Software: Same message as above.
Check MJ Version: You are currently running version -Dated 4/11/2009

V1.6
Same as above, except the following:

Display Existing SIP Info: No Data Found. Click 1st Button to get current info.

When I click 1st Button, which is Get Current SIP Info, I get the above message under v1.5.

I am running Windows XP.

Can someone help please? Thanks.

Zen,

Here's what works for me. Have Stroths utility (I used 1.6) ready to go.

Click on the MJ Softphone Menu (think it's the second button from the right on the top.
Select Restart
When MJ is showing the progress bar about 3/4 of the way across,
Click on Stroths utility to get SIP info.
It takes a little time, but don't do anything until you get the nice text box with the info or the little box that says SIP info not found.

It may take a couple times but this has been pretty reliable for me.

Let us know if this works for you. If not we'll try something else.

OTC

Oldtimercurt, you are great. It worked, I got all SIP, etc. info and used it as MagicDump has it on page 1 of this thread. However, I don’t have a dial tone. When I long into my PAP2, in the info section, under Line 1 Status, on the right next to Registration State: “Can’t connect to login server.”

I don’t know what could cause this. I am researching it, but if you could think of what may be causing this please let me know.

Thanks Oldtimercurt and MagicDump for your help.

Zen, double check your password entry. When I first tried setting up my PAP2T the other day, I got the same thing. I compared MagicDump's data page by page and couldn't find anything wrong. So I just reentered my password, saved and lo and behold it was registered.

OTC

I just tried that, but no luck. On line 1 for Display Name, User ID, and Auth ID, I am using my ProxyUserName (Exxxxxxxxxx01) this is my phone number +. For Password, I am using ProxyUserPassword, this is the combination of numbers and letters.

For User-Agent, I am using: MagicJack/1.80.466c (SJ Labs). I don't know if I should include the (SJ Labs) or not.

I am not making use of UserDomain=talk4free.com, I don’t know if I need to.

Both of the phone line LEDs are off on the PAP2T.

I am disclosing this so if I am doing something wrong, it can be brought to my attention.

Thanks again OTC.

OK, I got it to work! I have a dial tone and can make calls, but inbound calls go to VM. A couple of inbound calls came through, but all are going to VM.

What a day. It took me most of the day to configure it and it still doesn't work properly. But I have definitely made progress.

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: Select all

([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!)

I have used this Code in my PAP2T. I have to dial 1+ area code+number, even if I am calling a number in my area code. I like to know if something in this code can be changed so to make dialing 1 unnecessary. I have changed the (311) with my area code and it does not make any deference.

I have taken the (011!) out so I can make international calls.

Exactly what happens when you dial a 7 digit call? Exactly what does your Info screen say for your "Last Dialed Number"?

"...does not make any deference [sic]" gives nothing to work with.

Ok my bad, it works. Here is what happens. When I have the code exactly as you have suggested, I have to dial 1+area+number. When I change the (1311) to (1my area), 7 digits within my area works. If I dial area+number, I have to dial 1 first regardless of area. If I don’t dial 1, I hear the phone ring out but the phone I am calling does not ring, so I don’t know where the call is going.

I was used to not dialing 1 before using the PAP2T, so I am wondering if that can be accomplished, if not, I can get used to it.

I am having a problem with making international calls. I have removed the (011!) from the code and as soon as I dial the first (0) in an attempt to dial (011) to make an international call, I get a fast busy signal. I definitely need help with this as I make alot of international calls.

The Dial Plan will not work, and you will be forever dialing wrong numbers, if 10 digit dialing is made to coexist with 7 digit dialing in a Linksys/Sipura ATA. The reason is that there is just not enough space in the Dial Plan field to fit all of the legitimate 10 digit combinations while excluding the illegitimate ones. This is necessary in order to forward the digits to the SIP server automatically after the "right" number of digits is entered, rather than always waiting for the short interdigit timer to expire before forwarding the digits. POTS switches that permit some 10 digit dialing do so only for local Area Code overlay purposes, and they exclude certain combinations just for this reason.

You could always shunt the timer by putting # at the end of every dialed number, but wouldn't it be easier to put 1 in front than # behind?

As for international dialing, please remember that this Dial Plan is designed to be perfect. That does not mean free of errors; it means that there is exactly one Dial Plan element for any dialed digit sequence and that element either permits or prohibits that sequence. You have eliminated the 011 prohibition without replacing it with an appropriate 011 permission. Right now when you dial 0 the SPA knows immediately that it does not know how to process that digit sequence and gives you a Congestion Service Signal.

As I asked in the other thread when another member brought up this very same topic, are there any particular countries you want to permit (so you don't get stuck with a $100 Inmarsat bill because you misdialed a 10 digit number), or do you need to facilitate calling to any country or territory?

I agree it would be much easier to just dial 1 first.
As for the international calls, I like to know both, how to facilitate calling to any country and how to only permit calls to Germany (011-49-30-#) and Switzerland (011-41-31-#).

mberlant, since you know so much about this stuff as I have also seen your explanations in BroadVoice forum. Do you know how is a DID created? I always wanted to know, but haven’t found the answer.

The safest way to proceed is to add 01149xxx. for Germany and 01141xxx. for Switzerland to the Dial Plan. Don't forget the | separator, as appropriate. If you want to further limit calls to Berlin and Bern, just insert the City Codes in those strings.

There are two ways to unlock international dialing, one is safer and one is shorter. The shorter way is to add 011[2-9]xxx. to the Dial Plan. The safer way is to add <00:>011[2-9]xxx. to the Dial Plan. The second option requires you to dial an extra 00 in front of 011, just to help prevent accidentally dialing an expensive call. It also prevents household guests from running up your phone bill without your permission.

mberlant, in your Germany example you only have 3 xxx after the 01149. Why only 3? Do you not have to provide for every number? I did mine as 01149xxxxxxx and it seems to work. Does it matter?

Thanks for all you do.

OTC

It's just arbitrary, to prevent misdials from too-short numbers. German POTS numbers can vary in length from 9 digits to 15 digits, including the Country Code, depending upon population density and whether the exchange is serving a DID PABX. If the SPA's Dial Plan element is too short, misdials can occur more easily and, if too long, you won't be able to connect to someone who has a short phone number.

Even today there are just as many countries with mixed length or variable length phone numbers as with fixed length phone numbers. SPAs have a limit of about 4000 characters in the Dial Plan, which is not nearly enough to implement all of the rules I do know about, let alone the rules I don't know about.

I am using 2 MJs with my PAP2T, one on each line. When I have only one line activated it works fine, but when I activate both lines the ring on both lines sound different. When a call comes in on either line it sound like having a distinctive ring. My Dist Ring Setting is (NO) for User1 & User2.

Any idea what may be causing this

At the bottom of each User page is the setting for Default Ring. If they are both set to the same value the cadence of both lines will be the same. If the pitch of the two lines is different, it must be a feature of your telephone, since the PAP2 can only ring a phone with different cadences.

mberlant, thank you for all your help. Modifying my Dial Plan enable me to make international calls. However, changing the Default Ring Setting values for the Users did not change the ring pattern. I have also changed phones, but I continue to have a distinctive ring patter for both users. I am not sure if my IP PBX has anything to do with it, but for now I am happy with the results.

I'm able to make one or two calls and then I'm able to receive one or two calls and then the rest of the calls go to VM. I have set the register expires to 180 and i'm on a NY proxy. Is there anything else i can try so the calls dont go straight to VM? Please help

Please do not crosspost. Your reply is waiting in the other thread.

mberlant wrote:

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.

STUN is a waste of time and is unreliable. Just leave it alone...it's just one more failurepoint. I have an airlink wireless N router...and not using this stupid stun BS. I tried it...and guess what? It jacked my PAP up and wouldn't work. Turned it off...and guess what? It works.

mberlant wrote:You use it as a reference when programming your own ATA.

I have the same question with mjtricks post above. but I openned the file and change line 1 to my city name, EXXXO1, PASSWORD, and save. Now what am I going to do next? Thank for advance...

HI Oldtimercut I have same problem Zen, but I only get blink windows . Please help or any advice. I run vista, thank for advance..

oldtimercurt wrote:Zen,

Here's what works for me. Have Stroths utility (I used 1.6) ready to go.

Click on the MJ Softphone Menu (think it's the second button from the right on the top.
Select Restart
When MJ is showing the progress bar about 3/4 of the way across,
Click on Stroths utility to get SIP info.
It takes a little time, but don't do anything until you get the nice text box with the info or the little box that says SIP info not found.

It may take a couple times but this has been pretty reliable for me.

Let us know if this works for you. If not we'll try something else.

OTC

Godragonking wrote:

mberlant wrote:You use it as a reference when programming your own ATA.

I have the same question with mjtricks post above. but I openned the file and change line 1 to my city name, EXXXO1, PASSWORD, and save. Now what am I going to do next? Thank for advance...

The highlighted O should be a 0 <zero>

Has anyone been able to get 3-Way calling working. If I am on a call and then receive another placing one on hold, I am supposed to be able to do a 3 way call by punching in ## on MajicJack and disconnect it it using #*. Has anyone managed to do the same thing with the PAP2? I tried it but didn't seem to work. I have a Linksys PAP2-NA 3.1.22(LS) if that matters.

BTW, the info in this theed worked greet. Thanks.. I used spaconf and uploaded the settings file posted here. I did have to comment out a couple of things in the config file probably due to differences between our PAP2 versions, but it still worked.

Thanks...

Smee

I am using mj on spa-3102 but i am having a little problem, whenever i make an outgoing call using the phone connected to spa-3102 it takes about 11 sec before it even rings, i tried mj usb dongle with my computer and made the outgoing call and it rang in may 2 sec, anyone have any idea what setting in my spa-3102 ata needs to be changed, Thanks in advance

zahidniaz wrote:I am using mj on spa-3102 but i am having a little problem, whenever i make an outgoing call using the phone connected to spa-3102 it takes about 11 sec before it even rings, i tried mj usb dongle with my computer and made the outgoing call and it rang in may 2 sec, anyone have any idea what setting in my spa-3102 ata needs to be changed, Thanks in advance

It's due to your dial plan most likely. Probably due to a 7, 10 & 11 digit dialing rule. Search the forums for that.

Smee

MagicDump wrote:

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

Please help me to setup my PAp2t to work without computer, I tried to config for couple weeks, but I am lost and do not know what to do now. I download above pap2tmj.config and what section do I replace my SIP?And do I leave all section untouch? I very needed your help or advice...

Smee wrote:

zahidniaz wrote:I am using mj on spa-3102 but i am having a little problem, whenever i make an outgoing call using the phone connected to spa-3102 it takes about 11 sec before it even rings, i tried mj usb dongle with my computer and made the outgoing call and it rang in may 2 sec, anyone have any idea what setting in my spa-3102 ata needs to be changed, Thanks in advance

It's due to your dial plan most likely. Probably due to a 7, 10 & 11 digit dialing rule. Search the forums for that.

Smee

Thanks, it was the dial plan which was taking 11 sec to dial out i got the right dial plan from this thread and its working perfect now, Thanks A Lot

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: Select all

([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!)

you seem to be good at this dialing stuff! i am wanting to setup like a *123 or just 123 to call voice mail in the router is this possible? if so how do i do it please


Thanks
911pcdoc

MagicDump wrote:

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

Please help, I openned and changed line1 with mysipinfo.txt, then save overwrite the PAP2tMJ.config. I tried to read and follow so many times, but not much understand what to do next. I downloaded spaconf and Python 2.6. unziped and do know what to do next. Needed more help or step by step instructions to follow. Thank for advance..

Firmware Version: 3.1.22(LS)
Voice


Phone Adapter with 2 Ports for Voice-Over-IP


PAP2

Info
System
SIP
Provisioning
Regional
Line 1
Line 2
User 1
User 2
Advanced View (switch to basic view) User Login

System Information
DHCP: Enabled Current IP: 192.168.1.41
Host Name: LinksysPAP Domain: open dns
Current Netmask: 255.255.255.0 Current Gateway: 192.168.1.1
Primary DNS: 192.168.1.1
Secondary DNS:
Product Information
Product Name: PAP2-NA Serial Number: FH900EB42477
Software Version: 3.1.22(LS) Hardware Version: 0.03.4
MAC Address: 000F9A1A00D8 Client Certificate: Installed
Customization: Open
System Status
Current Time: 6/6/2009 19:07:36 Elapsed Time: 00:00:02
Broadcast Pkts Sent: 0 Broadcast Bytes Sent: 0
Broadcast Pkts Recv: 0 Broadcast Bytes Recv: 0
Broadcast Pkts Dropped: 0 Broadcast Bytes Dropped: 0
RTP Packets Sent: 0 RTP Bytes Sent: 0
RTP Packets Recv: 0 RTP Bytes Recv: 0
SIP Messages Sent: 1 SIP Bytes Sent: 581
SIP Messages Recv: 0 SIP Bytes Recv: 0
External IP:
Line 1 Status
Display Name: Exxxxxxxx01 User ID: Exxxxxxxx01
Hook State: On Registration State: Can't connect to login server
Last Registration At: 0/0/0 00:00:00 Next Registration In: 29 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
Call 1 Encoder: Call 2 Encoder:
Call 1 Decoder: Call 2 Decoder:
Call 1 FAX: Call 2 FAX:
Call 1 Type: Call 2 Type:
Call 1 Remote Hold: Call 2 Remote Hold:
Call 1 Callback: Call 2 Callback:
Call 1 Peer Name: Call 2 Peer Name:
Call 1 Peer Phone: Call 2 Peer Phone:
Call 1 Duration: Call 2 Duration:
Call 1 Packets Sent: Call 2 Packets Sent:
Call 1 Packets Recv: Call 2 Packets Recv:
Call 1 Bytes Sent: Call 2 Bytes Sent:
Call 1 Bytes Recv: Call 2 Bytes Recv:
Call 1 Decode Latency: Call 2 Decode Latency:
Call 1 Jitter: Call 2 Jitter:
Call 1 Round Trip Delay: Call 2 Round Trip Delay:
Call 1 Packets Lost: Call 2 Packets Lost:
Call 1 Packet Error: Call 2 Packet Error:
Call 1 Mapped RTP Port: Call 2 Mapped RTP Port:
Line 2 Status
Display Name: User ID:
Hook State: On Registration State: Offline
Last Registration At: Next Registration In:
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
Call 1 Type: Call 2 Type:
Call 1 Encoder: Call 2 Encoder:
Call 1 Decoder: Call 2 Decoder:
Call 1 FAX: Call 2 FAX:
Call 1 Remote Hold: Call 2 Remote Hold:
Call 1 Callback: Call 2 Callback:
Call 1 Peer Name: Call 2 Peer Name:
Call 1 Peer Phone: Call 2 Peer Phone:
Call 1 Duration: Call 2 Duration:
Call 1 Packets Sent: Call 2 Packets Sent:
Call 1 Packets Recv: Call 2 Packets Recv:
Call 1 Bytes Sent: Call 2 Bytes Sent:
Call 1 Bytes Recv: Call 2 Bytes Recv:
Call 1 Decode Latency: Call 2 Decode Latency:
Call 1 Jitter: Call 2 Jitter:
Call 1 Round Trip Delay: Call 2 Round Trip Delay:
Call 1 Packets Lost: Call 2 Packets Lost:
Call 1 Packet Error: Call 2 Packet Error:
Call 1 Mapped RTP Port: Call 2 Mapped RTP Port:

System Configuration
Restricted Access Domains:
Enable Web Server: Web Server Port:
Enable Web Admin Access: Admin Passwd:
User Password:
Internet Connection Type
DHCP:
Static IP: NetMask:
Gateway:
Optional Network Configuration
HostName: Domain:
Primary DNS: Secondary DNS:
DNS Server Order: DNS Query Mode:
Syslog Server: Debug Server:
Debug Level: Primary NTP Server:
Secondary NTP Server:

SIP Parameters
Max Forward: Max Redirection:
Max Auth: SIP User Agent Name:
SIP Server Name: SIP Reg User Agent Name:
SIP Accept Language: DTMF Relay MIME Type:
Hook Flash MIME Type: Remove Last Reg:
Use Compact Header: Escape Display Name:
RFC 2543 Call Hold: Mark All AVT Packets:
SIP Timer Values (sec)
SIP T1: SIP T2:
SIP T4: SIP Timer B:
SIP Timer F: SIP Timer H:
SIP Timer D: SIP Timer J:
INVITE Expires: ReINVITE Expires:
Reg Min Expires: Reg Max Expires:
Reg Retry Intvl: Reg Retry Long Intvl:
Reg Retry Random Delay: Reg Retry Long Random Delay:
Reg Retry Intvl Cap:
Response Status Code Handling
SIT1 RSC: SIT2 RSC:
SIT3 RSC: SIT4 RSC:
Try Backup RSC: Retry Reg RSC:
RTP Parameters
RTP Port Min: RTP Port Max:
RTP Packet Size: Max RTP ICMP Err:
RTCP Tx Interval: No UDP Checksum:
Stats In BYE:
SDP Payload Types
NSE Dynamic Payload: AVT Dynamic Payload:
INFOREQ Dynamic Payload: G726r16 Dynamic Payload:
G726r24 Dynamic Payload: G726r32 Dynamic Payload:
G726r40 Dynamic Payload: G729b Dynamic Payload:
NSE Codec Name: AVT Codec Name:
G711u Codec Name: G711a Codec Name:
G726r16 Codec Name: G726r24 Codec Name:
G726r32 Codec Name: G726r40 Codec Name:
G729a Codec Name: G729b Codec Name:
G723 Codec Name:
NAT Support Parameters
Handle VIA received: Handle VIA rport:
Insert VIA received: Insert VIA rport:
Substitute VIA Addr: Send Resp To Src Port:
STUN Enable: STUN Test Enable:
STUN Server: EXT IP:
EXT RTP Port Min: NAT Keep Alive Intvl:

Configuration Profile
Provision Enable: Resync On Reset:
Resync Random Delay: Resync Periodic:
Resync Error Retry Delay: Forced Resync Delay:
Resync From SIP: Resync After Upgrade Attempt:
Resync Trigger 1:
Resync Trigger 2:
Resync Fails On FNF:
Profile Rule:
Profile Rule B:
Profile Rule C:
Profile Rule D:
Log Resync Request Msg:
Log Resync Success Msg:
Log Resync Failure Msg:
Report Rule:
Firmware Upgrade
Upgrade Enable: Upgrade Error Retry Delay:
Downgrade Rev Limit:
Upgrade Rule:
Log Upgrade Request Msg:
Log Upgrade Success Msg:
Log Upgrade Failure Msg:
General Purpose Parameters
GPP A:
GPP B:
GPP C:
GPP D:
GPP E:
GPP F:
GPP G:
GPP H:
GPP I:
GPP J:
GPP K:
GPP L:
GPP M:
GPP N:
GPP O:
GPP P:

Call Progress Tones
Dial Tone:
Second Dial Tone:
Outside Dial Tone:
Prompt Tone:
Busy Tone:
Reorder Tone:
Off Hook Warning Tone:
Ring Back Tone:
Confirm Tone:
SIT1 Tone:
SIT2 Tone:
SIT3 Tone:
SIT4 Tone:
MWI Dial Tone:
Cfwd Dial Tone:
DND Dial Tone:
Holding Tone:
Conference Tone:
Secure Call Indication Tone:
Feature Invocation Tone:
Distinctive Ring Patterns
Ring1 Cadence: Ring2 Cadence:
Ring3 Cadence: Ring4 Cadence:
Ring5 Cadence: Ring6 Cadence:
Ring7 Cadence: Ring8 Cadence:
Distinctive Call Waiting Tone Patterns
CWT1 Cadence: CWT2 Cadence:
CWT3 Cadence: CWT4 Cadence:
CWT5 Cadence: CWT6 Cadence:
CWT7 Cadence: CWT8 Cadence:
Distinctive Ring/CWT Pattern Names
Ring1 Name: Ring2 Name:
Ring3 Name: Ring4 Name:
Ring5 Name: Ring6 Name:
Ring7 Name: Ring8 Name:
Ring and Call Waiting Tone Spec
Ring Waveform: Ring Frequency:
Ring Voltage: CWT Frequency:
Synchronized Ring:
Control Timer Values (sec)
Hook Flash Timer Min: Hook Flash Timer Max:
Callee On Hook Delay: Reorder Delay:
Call Back Expires: Call Back Retry Intvl:
Call Back Delay: VMWI Refresh Intvl:
Interdigit Long Timer: Interdigit Short Timer:
CPC Delay: CPC Duration:
Vertical Service Activation Codes
Call Return Code: Blind Transfer Code:
Call Back Act Code: Call Back Deact Code:
Cfwd All Act Code: Cfwd All Deact Code:
Cfwd Busy Act Code: Cfwd Busy Deact Code:
Cfwd No Ans Act Code: Cfwd No Ans Deact Code:
Cfwd Last Act Code: Cfwd Last Deact Code:
Block Last Act Code: Block Last Deact Code:
Accept Last Act Code: Accept Last Deact Code:
CW Act Code: CW Deact Code:
CW Per Call Act Code: CW Per Call Deact Code:
Block CID Act Code: Block CID Deact Code:
Block CID Per Call Act Code: Block CID Per Call Deact Code:
Block ANC Act Code: Block ANC Deact Code:
DND Act Code: DND Deact Code:
CID Act Code: CID Deact Code:
CWCID Act Code: CWCID Deact Code:
Dist Ring Act Code: Dist Ring Deact Code:
Speed Dial Act Code: Secure All Call Act Code:
Secure No Call Act Code: Secure One Call Act Code:
Secure One Call Deact Code: Conference Act Code:
Attn-Xfer Act Code: Modem Line Toggle Code:
Referral Services Codes:
Feature Dial Services Codes:
Vertical Service Announcement Codes
Service Annc Base Number:
Service Annc Extension Codes:
Outbound Call Codec Selection Codes
Prefer G711u Code: Force G711u Code:
Prefer G711a Code: Force G711a Code:
Prefer G723 Code: Force G723 Code:
Prefer G726r16 Code: Force G726r16 Code:
Prefer G726r24 Code: Force G726r24 Code:
Prefer G726r32 Code: Force G726r32 Code:
Prefer G726r40 Code: Force G726r40 Code:
Prefer G729a Code: Force G729a Code:
Miscellaneous
Set Local Date (mm/dd): Set Local Time (HH/mm):
Time Zone: FXS Port Impedance:
Daylight Saving Time Rule:
FXS Port Input Gain: FXS Port Output Gain:
DTMF Playback Level: DTMF Playback Length:
Detect ABCD: Playback ABCD:
Caller ID Method: FXS Port Power Limit:
Caller ID FSK Standard: Feature Invocation Method:
More Echo Suppression:


Line Enable:
Streaming Audio Server (SAS)
SAS Enable: SAS DLG Refresh Intvl:
SAS Inbound RTP Sink:
NAT Settings
NAT Mapping Enable: NAT Keep Alive Enable:
NAT Keep Alive Msg: NAT Keep Alive Dest:
Network Settings
SIP TOS/DiffServ Value: Network Jitter Level:
RTP TOS/DiffServ Value: Jitter Buffer Adjustment:
SIP Settings
SIP Port: SIP 100REL Enable:
EXT SIP Port: Auth Resync-Reboot:
Auth INVITE: Auth MWI:
SIP Proxy-Require: SIP Remote-Party-ID:
SIP GUID: SIP Debug Option:
RTP Log Intvl: Restrict Source IP:
Referor Bye Delay: Refer Target Bye Delay:
Referee Bye Delay: Refer-To Target Contact:
Sticky 183:
Call Feature Settings
Blind Attn-Xfer Enable: MOH Server:
Xfer When Hangup Conf: Conference Bridge URL:
Conference Bridge Ports:
Proxy and Registration
Proxy: Use Outbound Proxy:
Outbound Proxy: Use OB Proxy In Dialog:
Register: Make Call Without Reg:
Register Expires: Ans Call Without Reg:
Use DNS SRV: DNS SRV Auto Prefix:
Proxy Fallback Intvl: Proxy Redundancy Method:
Voice Mail Server: Mailbox Subscribe Expires:
Subscriber Information
Display Name: User ID:
Password: Use Auth ID:
Auth ID:
Mini Certificate:
SRTP Private Key:
Supplementary Service Subscription
Call Waiting Serv: Block CID Serv:
Block ANC Serv: Dist Ring Serv:
Cfwd All Serv: Cfwd Busy Serv:
Cfwd No Ans Serv: Cfwd Sel Serv:
Cfwd Last Serv: Block Last Serv:
Accept Last Serv: DND Serv:
CID Serv: CWCID Serv:
Call Return Serv: Call Back Serv:
Three Way Call Serv: Three Way Conf Serv:
Attn Transfer Serv: Unattn Transfer Serv:
MWI Serv: VMWI Serv:
Speed Dial Serv: Secure Call Serv:
Referral Serv: Feature Dial Serv:
Service Announcement Serv:
Audio Configuration
Preferred Codec: Silence Supp Enable:
Use Pref Codec Only: Silence Threshold:
G729a Enable: Echo Canc Enable:
G723 Enable: Echo Canc Adapt Enable:
G726-16 Enable: Echo Supp Enable:
G726-24 Enable: FAX CED Detect Enable:
G726-32 Enable: FAX CNG Detect Enable:
G726-40 Enable: FAX Passthru Codec:
DTMF Process INFO: FAX Codec Symmetric:
DTMF Process AVT: FAX Passthru Method:
DTMF Tx Method: DTMF Tx Mode:
FAX Process NSE: Hook Flash Tx Method:
FAX Disable ECAN: Release Unused Codec:
Dial Plan
Dial Plan:
Enable IP Dialing: Emergency Number:
FXS Port Polarity Configuration
Idle Polarity: Caller Conn Polarity:
Callee Conn Polarity:


Line Enable:
Streaming Audio Server (SAS)
SAS Enable: SAS DLG Refresh Intvl:
SAS Inbound RTP Sink:
NAT Settings
NAT Mapping Enable: NAT Keep Alive Enable:
NAT Keep Alive Msg: NAT Keep Alive Dest:
Network Settings
SIP TOS/DiffServ Value: Network Jitter Level:
RTP TOS/DiffServ Value: Jitter Buffer Adjustment:
SIP Settings
SIP Port: SIP 100REL Enable:
EXT SIP Port: Auth Resync-Reboot:
Auth INVITE: Auth MWI:
SIP Proxy-Require: SIP Remote-Party-ID:
SIP GUID: SIP Debug Option:
RTP Log Intvl: Restrict Source IP:
Referor Bye Delay: Refer Target Bye Delay:
Referee Bye Delay: Refer-To Target Contact:
Sticky 183:
Call Feature Settings
Blind Attn-Xfer Enable: MOH Server:
Xfer When Hangup Conf: Conference Bridge URL:
Conference Bridge Ports:
Proxy and Registration
Proxy: Use Outbound Proxy:
Outbound Proxy: Use OB Proxy In Dialog:
Register: Make Call Without Reg:
Register Expires: Ans Call Without Reg:
Use DNS SRV: DNS SRV Auto Prefix:
Proxy Fallback Intvl: Proxy Redundancy Method:
Voice Mail Server: Mailbox Subscribe Expires:
Subscriber Information
Display Name: User ID:
Password: Use Auth ID:
Auth ID:
Mini Certificate:
SRTP Private Key:
Supplementary Service Subscription
Call Waiting Serv: Block CID Serv:
Block ANC Serv: Dist Ring Serv:
Cfwd All Serv: Cfwd Busy Serv:
Cfwd No Ans Serv: Cfwd Sel Serv:
Cfwd Last Serv: Block Last Serv:
Accept Last Serv: DND Serv:
CID Serv: CWCID Serv:
Call Return Serv: Call Back Serv:
Three Way Call Serv: Three Way Conf Serv:
Attn Transfer Serv: Unattn Transfer Serv:
MWI Serv: VMWI Serv:
Speed Dial Serv: Secure Call Serv:
Referral Serv: Feature Dial Serv:
Service Announcement Serv:
Audio Configuration
Preferred Codec: Silence Supp Enable:
Use Pref Codec Only: Silence Threshold:
G729a Enable: Echo Canc Enable:
G723 Enable: Echo Canc Adapt Enable:
G726-16 Enable: Echo Supp Enable:
G726-24 Enable: FAX CED Detect Enable:
G726-32 Enable: FAX CNG Detect Enable:
G726-40 Enable: FAX Passthru Codec:
DTMF Process INFO: FAX Codec Symmetric:
DTMF Process AVT: FAX Passthru Method:
DTMF Tx Method: DTMF Tx Mode:
FAX Process NSE: Hook Flash Tx Method:
FAX Disable ECAN: Release Unused Codec:
Dial Plan
Dial Plan:
Enable IP Dialing: Emergency Number:
FXS Port Polarity Configuration
Idle Polarity: Caller Conn Polarity:
Callee Conn Polarity:

Call Forward Settings
Cfwd All Dest: Cfwd Busy Dest:
Cfwd No Ans Dest: Cfwd No Ans Delay:
Selective Call Forward Settings
Cfwd Sel1 Caller: Cfwd Sel1 Dest:
Cfwd Sel2 Caller: Cfwd Sel2 Dest:
Cfwd Sel3 Caller: Cfwd Sel3 Dest:
Cfwd Sel4 Caller: Cfwd Sel4 Dest:
Cfwd Sel5 Caller: Cfwd Sel5 Dest:
Cfwd Sel6 Caller: Cfwd Sel6 Dest:
Cfwd Sel7 Caller: Cfwd Sel7 Dest:
Cfwd Sel8 Caller: Cfwd Sel8 Dest:
Cfwd Last Caller: Cfwd Last Dest:
Block Last Caller: Accept Last Caller:
Speed Dial Settings
Speed Dial 2: Speed Dial 3:
Speed Dial 4: Speed Dial 5:
Speed Dial 6: Speed Dial 7:
Speed Dial 8: Speed Dial 9:
Supplementary Service Settings
CW Setting: Block CID Setting:
Block ANC Setting: DND Setting:
CID Setting: CWCID Setting:
Dist Ring Setting: Secure Call Setting:
Message Waiting: DND Activated:
Distinctive Ring Settings
Ring1 Caller: Ring2 Caller:
Ring3 Caller: Ring4 Caller:
Ring5 Caller: Ring6 Caller:
Ring7 Caller: Ring8 Caller:
Ring Settings
Default Ring: Default CWT:
Hold Reminder Ring: Call Back Ring:
Cfwd Ring Splash Len: Cblk Ring Splash Len:
VMWI Ring Splash Len:
VMWI Ring Policy:
Ring On No New VM:

Call Forward Settings
Cfwd All Dest: Cfwd Busy Dest:
Cfwd No Ans Dest: Cfwd No Ans Delay:
Selective Call Forward Settings
Cfwd Sel1 Caller: Cfwd Sel1 Dest:
Cfwd Sel2 Caller: Cfwd Sel2 Dest:
Cfwd Sel3 Caller: Cfwd Sel3 Dest:
Cfwd Sel4 Caller: Cfwd Sel4 Dest:
Cfwd Sel5 Caller: Cfwd Sel5 Dest:
Cfwd Sel6 Caller: Cfwd Sel6 Dest:
Cfwd Sel7 Caller: Cfwd Sel7 Dest:
Cfwd Sel8 Caller: Cfwd Sel8 Dest:
Cfwd Last Caller: Cfwd Last Dest:
Block Last Caller: Accept Last Caller:
Speed Dial Settings
Speed Dial 2: Speed Dial 3:
Speed Dial 4: Speed Dial 5:
Speed Dial 6: Speed Dial 7:
Speed Dial 8: Speed Dial 9:
Supplementary Service Settings
CW Setting: Block CID Setting:
Block ANC Setting: DND Setting:
CID Setting: CWCID Setting:
Dist Ring Setting: Secure Call Setting:
Message Waiting: DND Activated:
Distinctive Ring Settings
Ring1 Caller: Ring2 Caller:
Ring3 Caller: Ring4 Caller:
Ring5 Caller: Ring6 Caller:
Ring7 Caller: Ring8 Caller:
Ring Settings
Default Ring: Default CWT:
Hold Reminder Ring: Call Back Ring:
Cfwd Ring Splash Len: Cblk Ring Splash Len:
VMWI Ring Splash Len:
VMWI Ring Policy:
Ring On No New VM:

Registration State: Can't connect to login server

hours of hacking and it finally comes down to not have a port number on the proxy, thanks for all this info on this board you guys rock!!!!

When the phone rings it stutters can I make any adjustments to the settings to fix this??

Craig

My phone stutters the connection isnt clear when the phone is ringing or if im on a phone call its constantly cuts in and out on the call any adjustments i should be looking at??

craig

craigm1 wrote:When the phone rings it stutters can I make any adjustments to the settings to fix this??

Craig

Try going to your LINE 1 (Or Line 2 if that is where you configured your MJ) and change the Ring Waveform from Trapezoid to Sinusoid and see if that resolves your issue.

Smee

changed wave form still does it.

help!!!!!!!

craig

craigm1 wrote:changed wave form still does it.

help!!!!!!!

craig

What is your Ring1 Cadence setting? Have you also tried another phone.

Smee

have a cable modem and 4 lingo lines on the same connection as 2 majic jack lines with pap2, just tried one of mj lines with nothing else running and it sound perfect, I know with some of the carriers voip i have had there are setting on there website to lower bandwith requirements is there a way to do this directly in pap2??

craig

craigm1 wrote:Registration State: Can't connect to login server

Make sure you put in YourProxyServer:5070 in the proxy server line. Not just the YourProxyServer. The default port is 5060. This is what caused my "Can't connect to login server" error.

my sound is a little choppy when using multiple magic jack lines I saw someone on here say they configured the outbound proxy and it fixed it how would i conifgure this and is there a way to make the bandwith use of a pap2 lower for use with multiple lines

Thanks in advanced


craig

The PAP2TMJ.config file in the OP should be edited to replace

1.80.466c -> 1.80.484a (in two places)

and

proxy1 -> proxy01

Do people still have trouble with these changes applied, assuming one gets the new SIP password (if it's changed)?

I'm using the recommended settings we've discussed here. But I still have the same problem. Incoming Works. Outgoing does NOT.

How many people have incoming & outgoing successfully up?