Unofficial magicJack forum

macman4hire · Posted: Thu May 14, 2009 3:02 pm Post subject:

Glad to hear your setup is working and that you will help out other forum members.

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911pcdoc · Apprentice Joined: 12 May 2009 Posts: 20

question on dialing i think i want to set up dialing *123 for voicmail ie
*123 = xxx-xxx-xxxx do you know how i can do this??

anant

I have a RTP300 locked with Vonage service. Has anyone unlocked it to use with MJ? If this is already discussed in the forum I would appreciate if someone could guide me. Thanks in anticipation.

anant

ingenio · Newbie Joined: 11 Apr 2009 Posts: 9

UPDATTING DATE AND TIME

How do I DO?

I need to keep that information Update.

What Do I to need to do?

Thanks you.

911pcdoc · Apprentice Joined: 12 May 2009 Posts: 20

911pcdoc · Apprentice Joined: 12 May 2009 Posts: 20

ingenio · Newbie Joined: 11 Apr 2009 Posts: 9

YES I DIT IT, BUT NEVER UPDATE AUTOMATICALLY.

MAYBE I NEED SET OTHER THING.

PLEASE HELP ME.

THANKS YOU

911pcdoc · Apprentice Joined: 12 May 2009 Posts: 20

if the ntps are set and your reg time is set each time it resync it should get time !! time this might be odd but call your self form a cell or something it might kick the new time through the system!!

trek101 · Posted: Thu May 21, 2009 6:38 pm Post subject:

The reason why it takes 10 to 15 seconds for your call to connect is the interdigit long time by default is set for 10 seconds. Which means that the last digit dialed the ATA waits for about 10 seconds before forwarding the dialing information to the call server.

To change that just go to the Voice tab then Admin login link and then choose the Switch to advanced mode link. Click on the regional tab and go about a half way down. Change the Interdigit Long Timer to 5 seconds. That should shorten the delay on connecting your call after the last digit has been dialed...
_________________
That's my quarter's worth, but not enough to call someone who really cares at a pay telephone booth...

wirelessrat · Apprentice Joined: 09 May 2009 Posts: 10

Nice post. I got mine working, but I am expecting bugs. This should help me take care of any I suspect.

Thanks once again macman

otacon6530 · Newbie Joined: 25 Apr 2009 Posts: 8

sip user agent name?

it is cut off. what is the rest of it?

---------------------------------
I copied yours exactly except for the sip user agent name and I can not even get a dial tone.

on the information page it says registration Failed

macman4hire · Posted: Sun May 24, 2009 6:31 pm Post subject:

Just down load MagicDump's PAP2 configuration file and open the file using a text editor such as notepad. Find the part of the configuration that you can completely read in the screens shots and copy and paste that section into your RTP300 configuration. It clearly states this in the first post of this thread.

Good luck!

otacon6530 · Newbie Joined: 25 Apr 2009 Posts: 8

Sorry I have been at this for a long time and I must have over looked it.

No still does not work.

I do not even know where to go from here.

My registration State below Fails. Any suggestions?

Information
Product Name: rtp300 Serial Number: CJM00E956944
Software Version: 3.1.22 Hardware Version: 1.00.03
MAC Address: 0014BF910AC9 Client Certificate: Installed
System Status
Current Time: 1/1/2003 12:02:29 Elapsed Time: 00:02:29
RTP Packets Sent: 0 RTP Bytes Sent: 0
RTP Packets Recv: 0 RTP Bytes Recv: 0
SIP Messages Sent: 64 SIP Bytes Sent: 31251
SIP Messages Recv: 0 SIP Bytes Recv: 0
External IP: 67.236.70.150
Line 1 Status
Hook State: On Registration State: Failed
Last Registration At: 0/0/0 00:00:00 Next Registration In:
Message Waiting: Yes Call Back Active: No
Last Called Number: Last Caller Number:
Mapped SIP Port: 12986
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
Hook State: On Registration State: Not Registered
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:

otacon6530 · Newbie Joined: 25 Apr 2009 Posts: 8

oh I forgot to add this. When I pick up the phone I do not get a dial tone. I know it is connecting to the router because **** works

otacon6530 · Newbie Joined: 25 Apr 2009 Posts: 8

ok I tinkered with it and got it to work. I have no idea how but I am not complaining.

Is there any way to get the static in the background either gone or better than it is now?

I will try to mess with codecs and see if that helps.