magicJack and magicJack Plus Support, Reviews, FAQs and Tricks

esojmc wrote: Register String:
Exxxxxxxxxx01:[email protected]:5070

mberlant wrote:Also, you probably need to get rid of the port= line, or change it to 5060. In the case of a client definition like this, that number specifies your own listening port, not the server's. :5070 needs to stay in your register= line, as you have it, because that is the port you need to hail MJ on.

MagicDump, when you mean use userdomain=proxy1.miami.talk4free.com, are you suggesting that I switch my host information to use that server?

I feel so close and yet so far....It seems like there are about 10 different configurations, but yet, nothing seems to work just right.

joecanadian wrote:Now I don't use trickbox but my config is on freepbx and its pretty much the same..

Now I like to think I was one of the first to run mj on asterisks and have been running it for quite some time now as a small extra to my home phone system that allows me to make the odd longdistance call.

Its been great on and off for the last year.. I spent time on trying lots of configurations cause I had issues with my dtmf but this seems to be the best

Only change I made this month was my useragent from SJPhone to the latest changes.. i guess SJPhone was bought out by the MJ peoples
so ..
useragent=SJPhone
to
useragent=MagicJack/1.80.466c (SJ Labs)

I am not sure if the useragent sends this info and have not really tried to figure a way to check it

As I look at the settings today I dunno if I really need all the things like
auth=md5
canreinvite=no
insecure=very

But things seemed to work fine

here are my settings.. (been using them with out a hitch for more then a year now

Code: Select all

allow=ulaw
auth=md5
canreinvite=no
context=from-trunk
disallow=all
dtmfmode=rfc2833
fromuser=EXXXXXXXXXX01
host=216.234.69.8
insecure=very
nat=yes
port=5070
secret=SOMELONGALPH8NUMERICTHINGHERE
type=friend
useragent=MagicJack/1.80.466c (SJ Labs)
username=EXXXXXXXXXX01

enjoy

gorack wrote:See this is were i get confused, I have added this in trixbox using the gui.. but all these things about manually adding stuff..

Were do we add this?

zipsnet wrote:I have my box setup just like this but I get this:

<--- SIP read from UDP://216.234.79.8:5070 --->
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 66.XXX.XXX.XXX:5060;branch=z9hG4bK0b355bcc;rport
To: <sip:216.234.79.8:5070>
From: "Unknown"<sip:[email protected]>;tag=as775649f3
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: ENSR2.5.57.2-IS31
Content-Length: 0


Any Idea??

Thanks.

mberlant wrote:That field is not for authentication. That field is a token label that tells sip.conf what context within extensions.conf to send an incoming call to. The number you put behind the / at the end of the register= line is the stanza within that context to send an incoming call to.

In your example, an incoming call will be passed to

Code: Select all

[magicjack]
3115552368,1,.........
3115552368,2,.........

You then need to have written code in that location in extensions.conf (or have had some program do that for you) that tells Asterisk what to do with the inbound call.

elsmacko wrote:I was also having problems with not getting inbound to work. What did it for me was adding my phone number to the end of the registrations string

So after port

5070/1234567890 (Notice no E or 01 or anything added, just the 10 digit number). Some of my other settings in inbound are not exact same but it seems to be working, so I left them as is.

THanks! I was having to have 2 phones on my desk to use the MJ. Now I don't.

jjanton wrote:Hi,
I'm not sure if this is good practice ( not trying to hijack this thread)
but I'm having a similar issue I can receive in but not out
I've registered with phone number @ end and have also changed many of the settings in the trunk.
They are as follows

username=Exxxxxxxxxx01
fromuser=Exxxxxxxxxx01
dtmfmode=rfc2833
secret=my password
host=xxx.xxx.xxx.xxx
insecure=very
nat=yes
type=friend
qualify=2000
disallow=all
allow=ulaw
useragent=MagicJack/1.80.466c (SJ Labs)

username=EXXXxxxXXXX01
context=from-trunk
dtmfmode=rfc2833
secret=my secret
host=xxx.xxx.xxx.xxx
insecure=very
nat=yes
type=friend
port=5070
qualify=2000
allow=ulaw
disallow=all

registry:

Exxxxxxxxxx01:[email protected]:5070/mjnumber


I am getting a "TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 3)" error when dialing out using x-lite

any help would be greatly appreciated

Thanks!

pbudden wrote:Hi,

I have the same problem as "hollettster"

I have checked, double checked and re dumped the MJ password from MJ running in Windows.

zuul*CLI> sip show registry
Host Username Refresh State Reg.Time
proxy1.Atlanta.talk4free.com:5 EXXXXXXXXXX0 120 Auth. Sent
sipgate.com:5060 3xxxxxxx0 105 Registered Sun, 27 Sep 2009 11:47:24


After following the directions on the first page of this thread, here are my sanitized settings:

username=EXXXxxxXXXX01
secret=Super Secret 20 Char pw
type=friend
useragent=MagicJack/1.80.466c (SJ Labs)
qualify=2000
port=5070
nat=yes
host=216.234.78.8
fromuser=EXXXxxxXXXX01
dtmfmode=inband
insecure=very
context=from-pstn



Registration String:
EXXXxxxXXXX01:(Super secret PW)@proxy1.Atlanta.talk4free.com:5070

Running the sip debug I see this:

<------------->
--- (9 headers 0 lines) ---
Responding to challenge, registration to domain/host name proxy1.Atlanta.talk4free.com
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 216.234.78.8:5070:
REGISTER sip:proxy1.Atlanta.talk4free.com:5070 SIP/2.0
Via: SIP/2.0/UDP 66.xxx.xx.xxx:5060;branch=z9hG4bK78600869;rport
From: <sip:[email protected]>;tag=as1da70e8b
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 109 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="EXXXxxxXXXX01", realm="stratus.com", algorithm=MD5, uri="sip:proxy1.Atlanta.talk4free.com:5070", nonce="7e8d28544_09258", response="75d8e50ec8821e077c4efb65360dc674"
Expires: 120
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0


---

<--- SIP read from 216.234.78.8:5070 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 66.xxx.xx.xxx:5060;branch=z9hG4bK78600869;rport
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=as1da70e8b
Call-ID: [email protected]
CSeq: 109 REGISTER
User-Agent: ENSR2.5.48.6-IS1-RMRG0-RG900-EP376524
WWW-Authenticate: Digest nonce="1dd2b8cba_09272",realm="stratus.com",algorithm=MD5
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: REGISTER
Reliably Transmitting (NAT) to 216.234.64.8:5070:
OPTIONS sip:216.234.64.8:5070 SIP/2.0
Via: SIP/2.0/UDP 66.xxx.xx.xxx:5060;branch=z9hG4bK41c02f60;rport
From: "Unknown" <sip:[email protected]>;tag=as26643835
To: <sip:216.234.64.8:5070>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 27 Sep 2009 16:03:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

Am I missing something?

I tried this on two installs of TB.
One internal on my network, and the other with a direct connection to the Internet.

I am open to ideas if anyone has any?

Thanks

crackerjack wrote:
ADD /XXXxxxXXXX to your registration string (YOUR MAGICJACK DID NUMBER PRECEDED BY BACKSLASK)
restart asterisk

what is cli output when you type sip show registry and sip show peers?


good luck

Crackerjack

tvland wrote:I'm running trixbox 2.8.0.1 with asterisk 1.6.0.9

The settings seem to work as far as today. My problem is that even with my telephone number at the end of the register string, incoming calls are not routed by that DID.To make it work, I have to use the any DID rule and the incoming calls work. Without the any DID rule, asterisk gets the call but returns a not in service message. I tried just the number or the username with the E and incoming calls are still not routed by the DID. I tried just about every way I could think of.

Could this be something to do with mjproxy?

Any suggestions?

VaHam wrote:I haven't tried using the MJMD5 proxy but have you looked at the alternative method in the following thread? http://www.phoneservicesupport.com/magi ... t7243.html

tvland wrote:

VaHam wrote:I haven't tried using the MJMD5 proxy but have you looked at the alternative method in the following thread? http://www.phoneservicesupport.com/magi ... t7243.html

If you read that thread you'll see I've already tried that! There is another problem involved patching Asterisk in trixbox. Since the latest version is a patched Asterisk, I haven't been able to find the source code files to patch my version which is Asterisk 1.6.0.9-samy-r27.

crackerjack wrote:Found your problem right here...
[2010-10-03 05:48:33] NOTICE[2957] chan_sip.c: -- Registration for '[email protected]' timed out, trying again (Attempt #1)
Obviously you need to type faster, and learn some forum manners

markymark wrote:

crackerjack wrote:Found your problem right here...
[2010-10-03 05:48:33] NOTICE[2957] chan_sip.c: -- Registration for '[email protected]' timed out, trying again (Attempt #1)
Obviously you need to type faster, and learn some forum manners

what do you mean?

sabresfan wrote:

markymark wrote:

crackerjack wrote:Found your problem right here...
[2010-10-03 05:48:33] NOTICE[2957] chan_sip.c: -- Registration for '[email protected]' timed out, trying again (Attempt #1)
Obviously you need to type faster, and learn some forum manners

what do you mean?

http://www.youtube.com/watch?v=bVL3b1wKZQU