MagicJack Setup Guide For TrixBox CE

mykroft · Post by **mykroft** » Mon Oct 20, 2008 2:27 am

Updated 6/8/2009

DISCLAMER-----Using this guide can get your MJ account terminated – I take no responsibility for what could happen if you use this guide.

This is a guide that will help you configure your magicjack service to work with TrixBox CE 2.6.1.13. Please not that I had alot of help from a person who wants to remain anonymous - alot of this information was pulled together from alot of different sources.

This guide WILL NOT assist you in acquiring your SIP credentials. You can Google and find that information easily.

Load up your TrixBox CE GUI and go to PBX > PBX Settings > Trunks > Add SIP Trunk

General Settings -

Leave these settings blank for their defaults – you can not specify Outbound Caller ID as MJ blocks changing it. Also since magicJack is a single user device - if you set the channels to more than 1, your setup will not function correctly.

Outgoing Dial Rules -

Set these up as you need – I used 1NXXNXXXXXX only cause I don't have a local number available to me yet – and I also use a zap channel for local calls.

Outgoing Settings -

Trunk Name: MagicJack
PEER Details:
username=EXXXXXXXXXXX1
type=friend
secret=XXXXXXXXXXXXXXXXXXXX
qualify=2000
port=5070
nat=yes
host=XXX.XXX.XXX.XXX
fromuser=EXXXXXXXXXXX1
dtmfmode=inband
insecure=very
context=from-pstn

Incoming Settings:

BLANK

Registration: -

Register String: EXXXxxxXXXX01:XXXXXxxxxxXXXXXxxxxx@proxy1.<City>.talk4free.com:5070

Variable References: -

EXXXxxxXXXX01 = your MagicJack phone number, E on the front and 01 on the end
XXXXXxxxxxXXXXXxxxxx = your decrypted MagicJack password - 20 digit alpha/numeric
<City> = pick the closest point to you from the list below
XXX.XXX.XXX.XXX = numeric IP of the proxy address you selected
Nat = yes or no depending on your requirement

Here is a list of proxy servers that I know Of: -

proxy1.Atlanta.talk4free.com
proxy1.Austin.talk4free.com
proxy1.Baltimore.talk4free.com
proxy1.Boston.talk4free.com
proxy1.Chicago.talk4free.com
proxy1.Cleveland.talk4free.com
proxy1.Columbus.talk4free.com
proxy1.Dallas.talk4free.com
proxy1.Denver.talk4free.com
proxy1.Detroit.talk4free.com
proxy1.Houston.talk4free.com
proxy1.LasVegas.talk4free.com
proxy1.LosAngeles.talk4free.com
proxy1.Memphis.talk4free.com
proxy1.Miami.talk4free.com
proxy1.Minneapolis.talk4free.com
proxy1.Nashville.talk4free.com
proxy1.Newark.talk4free.com
proxy1.NewYork.talk4free.com
proxy1.Philadelphia.talk4free.com
proxy1.Phoenix.talk4free.com
proxy1.Pittsburgh.talk4free.com
proxy1.Portland.talk4free.com
proxy1.Sacramento.talk4free.com
proxy1.SaltLakeCity.talk4free.com
proxy1.SanDiego.talk4free.com
proxy1.SanFrancisco.talk4free.com
proxy1.Seattle.talk4free.com
proxy1.StLouis.talk4free.com
proxy1.Tampa.talk4free.com
proxy1.Washington.talk4free.com

After entering all the required information click submit changes and commit them to the server

Without the Context=from-pstn - asterisk would answer - but say the number was not in service

Setup your outbound routes as needed and order them they way you want them to be used.

By using the PBX status screen you should see:

Sip Registry
your selected proxy1 address in the Sip Registry = with status as registered – if not – check your userid and password in the trunk registry setting.

Sip peers
you should see MagicJack/EXXXxxxXXXX using your host ip selected and status of OK
should also see Inbound./EXXXxxxXXXX using your host ip selected and status of OK

You should now be able to send and receive calls via your new SIP service.

If you see any typos etc please notify me so I can keep this guide updated.

JohnODon · Post by **JohnODon** » Mon Oct 20, 2008 8:08 am

Can someone explain thsi to the ignorant please???

I know absolutely nothing about PBX...what are the high level benefits? Mailboxes? Multiple lines?

TIA,

John

mykroft · Post by **mykroft** » Mon Oct 20, 2008 3:01 pm

The PBX manager in TrixBox CE is a GUI editor for all the .conf files and alot more. You can take the above info and plug it is manually by editing the various .conf files (sip.conf, extensions.conf etc) and input the same infomation. It is just 100% easier to use the GUI editor that comes with TrixBox.

If you dont know what asterisk is - google is a good place to start for info to read.

Myk

oliverda · Post by **oliverda** » Mon Oct 20, 2008 5:24 pm

This is useless. I give up. I'll use MJ the legal way.

mykroft · Post by **mykroft** » Mon Oct 20, 2008 8:56 pm

oliverda wrote:This is useless. I give up. I'll use MJ the legal way.

Whats the problem? Its working perfectly for me right now - just spent several hours on the phone via TrixBox > MJ

Myk

Poo619 · Post by **Poo619** » Tue Oct 21, 2008 8:15 am

question for you mykroft. If I use Trix Box GUI in order to get the PBX working how I wanted to, can I then copy the .conf files over to an asterisk server I have running on a Fonera. I want to have * running in my house but I cant justify running a PC all day long for the limited amount of calls I make/receive. Basically what I am asking is if this will put it into standard asterisk form that any system running it will understand. I would do it by hand but I can not seem to grasp it as much as I would like to.

dan · Post by **dan** » Tue Oct 21, 2008 9:08 am

I realize this is an open forum. to share info about magicjack and to help new users out.

Perhaps what is being posted here is a little too much information. This has been a great service and I would hate to see MJ guys lock every thing down..

Maybe this topic should be taken off line? Just my two cents

Admins what do you think?

mykroft · Post by **mykroft** » Tue Oct 21, 2008 1:47 pm

What is the difference of getting Asterisk to with with magicjack vs getting 3rd party softphones working with magicjack. I dont see you hollaring to get those posts removed? My guide is just information on how to get it to work, it does not go about how to crack the magicjack arrount info that is required 1st - and that is not real easy for the average person.

All this information is out there already on these forums in bits and pieces, I just put it together in a coherent guide.

and FYI - this is the hacking forum for magicJack.......

dan wrote:I realize this is an open forum. to share info about magicjack and to help new users out.

Perhaps what is being posted here is a little too much information. This has been a great service and I would hate to see MJ guys lock every thing down..

Maybe this topic should be taken off line? Just my two cents

Admins what do you think?

mykroft · Post by **mykroft** » Tue Oct 21, 2008 1:53 pm

Poo619 wrote:question for you mykroft. If I use Trix Box GUI in order to get the PBX working how I wanted to, can I then copy the .conf files over to an asterisk server I have running on a Fonera. I want to have * running in my house but I cant justify running a PC all day long for the limited amount of calls I make/receive. Basically what I am asking is if this will put it into standard asterisk form that any system running it will understand. I would do it by hand but I can not seem to grasp it as much as I would like to.

On a non PBXconf (which is a fork of FreePBX) setup, the info would go into sip.conf and extensions.conf - the PBX manager splits this information up into different files which is #included <filename> into the main .conf files,

So if your asterisk is not using freePBX and are using amportal - I would suggest taking a look at freePBX - it can be installed into any vanilla asterisk setup - and its free. freepbx.org

dan · Post by **dan** » Tue Oct 21, 2008 4:20 pm

I am paying for trunking through Vitelity and use MJ more or less to play around with. After MJ blocked PAI then I pretty much stoped using the device. so its not a huge deal if they lock this down for me. However dont be mad if they do..

you may want to consider adding

useragent = MagicJack/1.80.447a (SJ Labs)
via sip_general_custom.conf

also may want to change the To: fields in the regsiter/invtie sip messaing to match up..

CallerID Superfecta by nerd vittles is a great application for MJ

Dan

mykroft · Post by **mykroft** » Tue Oct 21, 2008 7:04 pm

User agent I was looking into and planing on adding it

Changing the To: fields in the register/invite msgs - not sure how to do that - any hints on where to look?

Myk

dan wrote:I am paying for trunking through Vitelity and use MJ more or less to play around with. After MJ blocked PAI then I pretty much stoped using the device. so its not a huge deal if they lock this down for me. However dont be mad if they do..

you may want to consider adding

useragent = MagicJack/1.80.447a (SJ Labs)
via sip_general_custom.conf

also may want to change the To: fields in the regsiter/invtie sip messaing to match up..

CallerID Superfecta by nerd vittles is a great application for MJ

Dan

jose.miami · Post by **jose.miami** » Tue Oct 21, 2008 11:19 pm

MagicJack/1.80.451b (SJ Labs), is the current SIP User Agent version

momo · Post by **momo** » Wed Oct 22, 2008 4:13 am

jose.miami wrote:MagicJack/1.80.451b (SJ Labs), is the current SIP User Agent version

doesn't this depend on the version mj software being used on the original mj computer?

if it's not the same on the original mj 'puter , won't there be a conflict when the dongle is plugged back in?

mykroft · Post by **mykroft** » Wed Oct 22, 2008 5:16 am

magicJack's software auto updates itself - so everyone should be running the newest/current version.

momo · Post by **momo** » Wed Oct 22, 2008 5:29 am

mykroft wrote:magicJack's software auto updates itself - so everyone should be running the newest/current version.

mine hasn't updated on my mj computer yet
it's still one or two versions back.

is anyone exactly sure it requires the actual latest version in the user agent?

after all, if you don't constantly keep the dongle plugged in, and just use it now and then, the service doesn't stop if you don't let the software update.

mine hasn't updated in the time i had it... it's not always plugged in yet, my system isn't fully set up for it... so... just wondering here if they compare the actual dongle usage version, to any other source usage version #

Thunder · Post by **Thunder** » Wed Oct 22, 2008 5:47 am

Thats the thing with TCs if you don't catch the update and do a comit it won't take the update. Oh and a new one is comming soon.

momo · Post by **momo** » Wed Oct 22, 2008 5:53 am

Thunder wrote:Thats the thing with TCs if you don't catch the update and do a comit it won't take the update. Oh and a new one is comming soon.

but it still doesn't say ...
is anyone exactly sure it requires the actual latest version in the user agent?

(... the service doesn't stop if you don't let the software update.)

just wondering here if they compare the actual dongle usage file version, to any other source usage file version #
giving them the "red flag" of different sources

oliverda · Post by **oliverda** » Sun Oct 26, 2008 8:47 am

[quote="mykroft"][quote="oliverda"]This is useless. I give up. I'll use MJ the legal way.[/quote]

Whats the problem? Its working perfectly for me right now - just spent several hours on the phone via TrixBox > MJ

Myk[/quote]

I put all the settings and nothing worked. I got the 20 character password. i'm wondering what i'm doing wrong. :?. Can anyone tell me what else needs to be done?. This is the error i'm getting.

MagicJack/EXXXXXXXXXX01 192.168.1.XXX 5070 UNREACHABLE
Inbound /EXXXXXXXXXX01 192.168.1.XXX N 5070 UNREACHABLE

mykroft · Post by **mykroft** » Sun Oct 26, 2008 2:25 pm

you did not follow instructions above on what to use for the host ip

look again, it does work

oliverda · Post by **oliverda** » Sun Oct 26, 2008 10:22 pm

I got it dude.:) I can send calls but i'm not able to receive calls.

mykroft · Post by **mykroft** » Mon Oct 27, 2008 1:56 am

double post

mykroft · Post by **mykroft** » Mon Oct 27, 2008 1:59 am

oliverda wrote:I got it dude. I can send calls but i'm not able to receive calls.

if your using a host=192.x.x.x address - that is incorrect

host=XXX.XXX.XXX.XXX is the numeric IP of the proxy address you selected

so if you are using a proxy of: proxy1.SaltLakeCity.talk4free.com

your host ip would be 67.108.236.70

mykroft · Post by **mykroft** » Mon Oct 27, 2008 2:01 am

Code: Select all

trixbox*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status               
magicjack/Exxxxxxxxxx01    67.88.208.198        N      5070     OK (104 ms)           
inbound/Exxxxxxxxxx01      67.88.208.198        N      5070     OK (104 ms)           
1000                       (Unspecified)    D   N      0        UNKNOWN              
3 sip peers [Monitored: 2 online, 1 offline Unmonitored: 0 online, 0 offline]

is what my sip show peers spits out

mykroft · Post by **mykroft** » Mon Oct 27, 2008 2:02 am

oliverda wrote:I got it dude. I can send calls but i'm not able to receive calls.

also, what is your inbound routing set to?

oliverda · Post by **oliverda** » Mon Oct 27, 2008 7:12 pm

Sip Peers
--------------------------------------------------------------------------------

Name/username Host Dyn Nat ACL Port Status

MagicJack/EXXXXXXXXXX01 67.91.233.134 5070 OK (73 ms)
Inbound /EXXXXXXXXXX01 67.91.233.134 N 5070 OK (70 ms)
18 sip peers [Monitored: 7 online, 9 offline Unmonitored: 2 online, 0 offline]

My inbound routing set to an extension. How the inbound routes should be?
I have more trunks working on the same pbx. Do you think that might be the problem? I'm using MJ for outbound mainly, but also I would like to have the inbound working properly.

mykroft · Post by **mykroft** » Tue Oct 28, 2008 2:54 pm

well,

you can route it like any other voip sip trunk, so if you have other - you can just add it to the group so it will get answered like normal with the others

dan · Post by **dan** » Wed Oct 29, 2008 7:40 am

You should post your trunk settings. be specefic on inbound route set up. I am sure if you follow the initial post on this thread it will work.

BTW anyone messed with the URI or TO: fields in the Invite?

dslab · Post by **dslab** » Tue Dec 02, 2008 7:17 pm

Is it still possible to setup MJ with trixbox CE?

I've followed the instructions here but without any success. Getting "All circuits are busy now" when trying to dial out. When trying to call in, it just says the extension is not available.

The status are "ok" for the inbound and outbound under sip registry (I am viewing this using the webgui, and it seems that sip registry and sip peers are reversed compared to what was described in this thread).

It won't register under Sip peer:

proxy1.<city>.talk4free.com:5070 EXXXXXXXXXXX1 120 Request Sent

mykroft · Post by **mykroft** » Wed Dec 03, 2008 3:36 pm

did you replace <city> with the closest city to you from the list posted?

And use the numerical ip where specified to match what city you used?

dslab · Post by **dslab** » Wed Dec 03, 2008 4:13 pm

Yes I did, I even tried different cities. But all give the same result. I used the ping command to find the actual IP of the proxy. I'm also forwarding Port 5070 to the trixbox. Don't know what else it could be.

mykroft · Post by **mykroft** » Thu Dec 04, 2008 3:48 pm

please post and edit out private info of your

sip show registry

from the asterisk cli

dslab · Post by **dslab** » Sat Dec 06, 2008 5:46 am

I thought the problem might be because I was using a older version of Trixbox, so I installed Trixbox 2.6.1.13, however getting the same result.

Here is the sip show registry from CLI:

Code: Select all


Trixbox*CLI> sip show registry
Host                            Username       Refresh State                Reg.              Time
proxy1.LosAngeles.talk4free.co  EXXXxxxXXXX0       120 Unregistered

As you can see the host and username are both truncated. I don't know if just cuts off because there is no room to display, or it actually got cut off and didn't send the whole string out.

mykroft · Post by **mykroft** » Sat Dec 06, 2008 7:43 pm

Your sip is not registering...

check your username and password - one of them is incorrect

dslab · Post by **dslab** » Mon Dec 08, 2008 7:03 am

Finally found the problem. I'd set an incorrect DNS server. Found that out when I realized I couldn't download any upgrade for Freepbx. Everything worked after changing that to a working DNS server IP.

I switch back to the other HDD with the older version of Trixbox (Version 2.2.4) to see if that was also the problem. However for that version it still wouldn't work. Maybe it doesn't allow registering using port 5070. Not a big deal, going to stick with the new one anyway.

Now the choice would be to continue on with Trixbox or try PBX in a Flash.

dslab · Post by **dslab** » Tue Dec 09, 2008 6:32 am

dan wrote: you may want to consider adding

useragent = MagicJack/1.80.447a (SJ Labs)
via sip_general_custom.conf

also may want to change the To: fields in the regsiter/invtie sip messaing to match up..

CallerID Superfecta by nerd vittles is a great application for MJ

What does adding the line

"useragent = MagicJack/1.80.447a (SJ Labs)"

in sip_general_custom.conf do?

crackerjack · Post by **crackerjack** » Tue Dec 09, 2008 12:26 pm

dslab wrote:
dan wrote: you may want to consider adding

useragent = MagicJack/1.80.447a (SJ Labs)
via sip_general_custom.conf

also may want to change the To: fields in the regsiter/invtie sip messaing to match up..

CallerID Superfecta by nerd vittles is a great application for MJ
What does adding the line

"useragent = MagicJack/1.80.447a (SJ Labs)"

in sip_general_custom.conf do?

Makes it appear to the YMAX servers as if the call is coming from a MJ device

magicjack_I_luv_you · Post by **magicjack_I_luv_you** » Tue Dec 09, 2008 5:26 pm

I can place outbound,but not inbound,failed authentication.

Does anybody have a suggestion? thanks in advance...

_______________________________________________________
I am registered.

*below while doing a tail -f /var/log/asterisk and then making an inbound call i get this username mismatch error, then failed authentication. I observe the second line "<sip:xxxXXXyyyy@..." does not have but 10 digits like i'm not seeing the 01 on the end but i'm not sure if that is pertinent. in the wireshark logs i see this "To: <sip:[email protected]:5060>", not sure why i should be seeing 5060. I am using gizmo i know it uses 5060 (well i think i know), but even when i delete its trunk it still shows the same errors. also, not sure if this is worth mentioning but am using 7910 on my tribox so had to do some sccp fanagaling. but did not see anything in the files i had to create and place in tftp to cause any issues.

while googleing about the "username mismatch" below, i was getting some ideas on checking the context values..

______________________________________________________
.....i am registered.......

trixbox*CLI> sip show registry
Host Username Refresh State Reg.Time
proxy1.brazil.talk4free.com:5 ExxxXXXyyyy01 105 Registered Tue, 09 Dec 2008 15:14:01

_______________________________________________________
......tail -f /var/log/asterisk/full

[WARNING[3733] chan_sip.c: username mismatch, have <MagicJack>, digest has <ExxxXXXyyyy01>

NOTICE[3733] chan_sip.c: Failed to authenticate user <sip:[email protected]:5070>;tag=11c22e33-co1122-INS111

_______________________________________________________

...below capture from wireshark of transaction.....

INVITE sip:[email protected]:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 222.333.77.7:5070;branch=11111111111111111111111-22222222-0

Max-Forwards: 70

Contact: <sip:[email protected]:5070>

To: <sip:[email protected]:5060> <<<why is this 5060?

From: <sip:[email protected]:5070>;tag=111111111-co2222222222

Call-ID: 11111111112222222222333333ens.com

CSeq: 515151 INVITE

Content-Type: application/sdp

Date: Tue, 09 Dec 2008 06:38:33 GMT

Supported: 100rel

User-Agent: asdfadsfasdfasdfasdfasdfasdfafsd

Content-Length: 279

v=0

o=- 555555555 555555555 IN IP4 222.333.77.7

s=ENSResip

c=IN IP4 12.12.12.12

t=0 0

m=audio 17512 RTP/AVP 8 0 18 3 101

a=fmtp:101 0-15

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=sendrecv

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP 222.333.77.7:5070;branch=11111111111111111111111-22222222-0;received=222.333.77.7

From: <sip:[email protected]:5070>;tag=111111111-co2222222222

To: <sip:[email protected]:5060>;tag=666666666666 <<<why is this 5060?

Call-ID: 11111111112222222222333333ens.com

CSeq: 515151 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="8888888d"

Content-Length: 0

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP 222.333.77.7:5070;branch=11111111111111111111111-22222222-0;received=222.333.77.7

From: <sip:[email protected]:5070>;tag=111111111-co2222222222

To: <sip:[email protected]:5060>;tag=666666666666

Call-ID: 11111111112222222222333333ens.com

CSeq: 515151 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="8888888d"

Content-Length: 0

ACK sip:[email protected]:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 222.333.77.7:5070;branch=11111111111111111111111-22222222-0

To: <sip:[email protected]:5060>;tag=666666666666

From: <sip:[email protected]:5070>;tag=111111111-co2222222222

Call-ID: 11111111112222222222333333ens.com

CSeq: 515151 ACK

Content-Length: 0

INVITE sip:[email protected]:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 222.333.77.7:5070;branch=yyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyy

Max-Forwards: 70

Contact: <sip:[email protected]:5070>

To: <sip:[email protected]:5060>

From: <sip:[email protected]:5070>;tag=111111111-co2222222222

Call-ID: 11111111112222222222333333ens.com

CSeq:515153 INVITE

Content-Type: application/sdp

Date: Tue, 09 Dec 2008 06:38:33 GMT

Proxy-Authorization: Digest username="ExxxXXXyyyy01",realm="asterisk",nonce="8888888d",uri="sip:s@my_external_IP:19859;user=phone",response="hhhhhhhhhhhhdddddddddddddfffffff",algorithm=MD5

Supported: 100rel

User-Agent: asdfadsfasdfasdfasdfasdfasdfafsd

Content-Length: 279

v=0

o=- 555555555 555555555 IN IP4 222.333.77.7

s=ENSResip

c=IN IP4 12.12.12.12

t=0 0

m=audio 17512 RTP/AVP 8 0 18 3 101

a=fmtp:101 0-15

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=sendrecv

SIP/2.0 403 Forbidden

Via: SIP/2.0/UDP 222.333.77.7:5070;branch=yyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyy;received=222.333.77.7

From: <sip:[email protected]:5070>;tag=111111111-co2222222222

To: <sip:[email protected]:5060>;tag=666666666666

Call-ID: 11111111112222222222333333ens.com

CSeq:515153 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Content-Length: 0

SIP/2.0 403 Forbidden

Via: SIP/2.0/UDP 222.333.77.7:5070;branch=yyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyy;received=222.333.77.7

From: <sip:[email protected]:5070>;tag=111111111-co2222222222

To: <sip:[email protected]:5060>;tag=666666666666

Call-ID: 11111111112222222222333333ens.com

CSeq:515153 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Content-Length: 0

ACK sip:[email protected]:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 222.333.77.7:5070;branch=yyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyyy

To: <sip:[email protected]:5060>;tag=666666666666

From: <sip:[email protected]:5070>;tag=111111111-co2222222222

Call-ID: 11111111112222222222333333ens.com

CSeq:515153 ACK

Content-Length: 0

proccw · Post by **proccw** » Wed Dec 17, 2008 9:09 pm

I'm running a Trixbox with MJ and it works great, but with dynamic IP addressing on occasion my IP address will change. Will I need to change my config every time my IP address changes? Or is there a way to automate this? I am using no-ip.com for dynamic dns and would love to be able to put in my domain name instead of ip address in the configs.

dslab · Post by **dslab** » Thu Dec 18, 2008 5:39 pm

proccw wrote:I'm running a Trixbox with MJ and it works great, but with dynamic IP addressing on occasion my IP address will change. Will I need to change my config every time my IP address changes? Or is there a way to automate this? I am using no-ip.com for dynamic dns and would love to be able to put in my domain name instead of ip address in the configs.

I am newbie with Asterisk, so I not that knowledgeable about this. But I think to do that you would place the following in your sip_custom.conf:

Code: Select all

externhost=myserver.dyndns.org
localnet=192.168.0.0/255.255.255.0      (NOTE: The first 3 octets need to match your private IP addresses!)

I think there are other methods to do this as well, you can read here:

http://knol.google.com/k/ward-mundy/pbx ... he_Install

Like I said, I am a newbie at this as well, so someone please correct me if I am wrong. Thanks.

dslab · Post by **dslab** » Thu Dec 18, 2008 5:45 pm

BTW, I am also unable to receive inbound call for the Magicjack number right now. Really not sure what I've done wrong. I've setup a "any DID/any CID" Inbound route. So I didn't think that was the problem.

I mostly use Magicjack for outbound calls, however since I can't change the caller ID for Magicjack, some people would call me back to the magicjack number. When that happens, they won't get through.

Any help would be appreciated.

zmenchhofer · Post by **zmenchhofer** » Sat Jan 10, 2009 5:54 pm

I can use OUTbound... but INbound calls dont work... any ideas?

crackerjack · Post by **crackerjack** » Sun Jan 11, 2009 1:34 am

zmenchhofer wrote:I can use OUTbound... but INbound calls dont work... any ideas?

Pls post your peer details for your trunk.

What version astx distro are you using?

Crack

zmenchhofer · Post by **zmenchhofer** » Sun Jan 11, 2009 1:08 pm

PEER:
host=proxy1.nashville.talk4free.com
username=xxx
fromuser=xxx
dtmfmode=inband
secret=XXX
type=friend
port=5070
qualify=2000

USER Details:
context=frommagicjack
username=xxx
fromuser=xxx
authuser=xxx
type=friend
secret=xxx
port=5070
nat=yes
insecure=very
host=proxy1.nashville.talk4free.com
fromdomain=sip.magicjack.com
dtmfmode=inband
qualify=2000
canreinvite=no
allow=all

Register String:

xxx:[email protected]:5070

benderbb · Post by **benderbb** » Mon Jan 12, 2009 1:44 am

No inbound here either. I show successful registration and have the inbound route setup any to extension. When I call from my cell, I just get hung up on immediately no message no ringing no anything. I have double checked the trunk settings and everything looks right to me. Any ideas? P.S. I added the line to make it look like the reg was coming from mj and that didnt help. I also noticed my dns under network settings was incorrectly reported as my gateway/router so I cahnged that to use opendns. I also cannot receive package updates for some reason.(seems related to a timeout setting though...prolly seperate issue.)

proccw · Post by **proccw** » Tue Jan 13, 2009 7:00 am

Don't use FQDN in your User, Peer or Registration. Use IP address only. Dynamic DNS doesn't matter in the configs, shouldn't be pointing to your own network. Not sure why I did that but there you go.

esojmc · Post by **esojmc** » Wed Jan 14, 2009 12:43 pm

OK, I have entered my MJ SIP credentials into TB. The trunk seem to register and I can make outgoing calls over MJ. Problem is that incoming calls doesn't even reach my TB. I dial my MJ number with my cell phone I get the following message (from Verizon) - "Welcome to Verizon Wireless. Your call cannot be connected as dialed...". I look at Asterisk logs and the call doesn't even reach my TB.

I plug the same information into a X-Lite softphone and I am able to receive and place calls.

Following is my TB setup:

Outgoing Settings
PEER Details:
username=Exxxxxxxxxx01
fromuser=Exxxxxxxxxx01
dtmfmode=inband
secret=PASSWORD
host=67.88.10.198
type=friend
port=5070
qualify=2000

Incoming Settings
User Details:
username=Exxxxxxxxxx01
type=friend
secret=PASSWORD
port=5070
nat=yes
insecure=very
host=67.88.10.198
fromuser=Exxxxxxxxxx01
dtmfmode=inband
qualify=2000

Register String:
Exxxxxxxxxx01:[email protected]:5070

Any help appreciated - Thanks.

crackerjack · Post by **crackerjack** » Wed Jan 14, 2009 1:00 pm

dump trixbox and go to nerdvittles.com and get pbx-in-a flash. Much more stable and reliable Asterisk distro than trixbox and you get a fully functional industry standard version of freepbx. You can easily migrate your existing settings and leave trixbox in the dustbin of hostory.

Check your PM

mykroft · Post by **mykroft** » Wed Jan 14, 2009 4:58 pm

Trixbox works just fine - yes they forked FreePBX, but there is nothing technically wrong with that.

I have a trixbox and a PBXiaf both running this config with no problems.

benderbb · Post by **benderbb** » Fri Jan 16, 2009 12:09 am

@mykroft... Since your setup is working, could you speculate as to why our inbound is not. I am thinking that my router may not be sip aware perhaps. however, I have ports 5070 and 5060 forwarded. Outbound calls work great. Am I missing any ports that need to get forwarded? My trixbox states:

MagicJack/Exxxxxxxxxx01 71.5.113.6 5070 OK (57 ms)
Inbound/Exxxxxxxxxx01 71.5.113.6 N 5070 OK (140 ms)
200/200 192.168.1.112 D N 5060 OK (43 ms)
3 sip peers [Monitored: 3 online, 0 offline Unmonitored: 0 online, 0 offline]

Any thoughts would be appreciated.

mykroft · Post by **mykroft** » Fri Jan 16, 2009 12:31 am

I will take a look at mine and double check that inbound is still working tonight when I get home from work.

I dont get much inbound on it, mainly outound.

Will post results.

Myk

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I can place outbound,but not inbound,failed authentication.

TrixboxCE with dynamic DNS

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