Topic: Getting SME Sail to work with budgetphone.nl help

[jreijsenbach]

Hi,

If anyone can help me get my test setup to work I would be very happy!!! 
I've been trying on and off to get Sail to work for me without a complete success for some time.
I guess it should be easy enough but I just seem to overlook something every time.

Here is the setup:
SME server/gateway => port 5060 opened to => SME testserver serveronly + Sail installed => Snom 190 + SPA 3000

Goal
I am looking to setup my testserver to do inbound and outbound calls on (initially just) one SIP account.
The account is with budgetphone.nl Voip provider.

Usefull information?
I found following information for asterisk servers in combination with budgetphone.nl:

---- extensions.conf ---------
[general]
[globals]

[extensions]
exten => _XX,1,Dial(SIP/${EXTEN},13)
exten => _XX,n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy,unavail)
exten => _XX,n(unavail),VoiceMail(10@default,u)
exten => _XX,n,Hangup()
exten => _XX,n(busy),VoiceMail(10@default,b)
exten => _XX,n,Hangup()
exten => 900,1,VoiceMailMain()

[toll-access]
include => extensions

exten => _0XXXX.,1,Dial(SIP/${EXTEN}@budgetphone)

[incoming]
include => extensions
exten => s,1,Goto(25,1)
exten => 31717111111,1,Goto(25,1)

---------- slip.conf -------------
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
svrloookup=no

register => 31number@budgetphone.nl:password:31number@sip.budgetphone.nl/31number

[budgetphone]
nat=no
type=peer
host=sip.budgetphone.nl
fromuser=31number
username=31number
secret=password
qualify=no
fromdomain=budgetphone.nl
context=incoming
insecure=very

[25]
type=friend
context=toll-access
host=dynamic
callerid="Username" <10>
canreinvite=no
disallow=all
allow=gsm
allow=alaw
qualify=no
secret=password

Any help with this would be greatly appreciated
It would be very nice if I could get it working since I know several people who would be
very happy with a howto.

With kind regards,

Jan

[SARK devs]

hello

First of all...  SIP/RTP calls use three separate ports...  5060 (UDP) is the control port (SIP) but the conversation is actually carried over RTP which can be any 2 ports between 10000 and 20000 (in asterisk's case).  They are chosen at random at call start-up.  Now, if your carrier is using Session Border Control and Symmetrical RTP then you may get away with just having port 5060 open and forwarded.  However, it is more usual to also have 10000-20000 open and forwarded to your asterisk server.  Otherwise you may get one-way sound or no sound.

You will need to create a carrier definition for budgetphone.nl.  There is a howto on the docs site here...

http://www.selintra.com/docs/cgi-bin/view/Main/DocChapter259

You will also need to ensure that you provide a registration string (the form is username:password@carrier-uri/username)

Finally, in your sip trunk you will need to provide the correct DID exactly as it will arrive from the carrier.  This will probabaly be of the form 31nnnnnnnnnnnnn (assuming budgetphone deliver numbers in E164 format).

Kind Regards

 

[wifi]

Jan,

I've got SME working fine (for some time now) with Budgetphone.nl  (i've even got 3 n's with them)
As soon as I have some more time this week, I will make a small howto. That is... If it's still needed.

[jreijsenbach]

Hi Wifi,

A small howto would be of great help indeed and much appreciated. I suspect it would be the shortest
route to the solution. Thanks in advance!

Hi Selintra,

The phones work directly with my provider no problem. Even if port 5060 isn't opened. So no firewall problem?
Ik did make a carrier definition and all and it registers the server no problem. The DID number I would have to
double check though I think I added that one correctly.

I hope wifi has time to write the short howto. I suspect it will be the quikest way to find out where I am making
the mistake(s).

Thanks for your ultra fast response! It amazes me everytime how fast you answer questions on this forum. 

Thanks both for your responses,

Looking forward to the howto.

With Kind regards,

Jan

[jreijsenbach]

bump.....

Hello Selintra,

Have made carrier definition. Checked the DiD. Alle setup correctly as far as I can see.

Am i right in assuming that the server only SME with SAIL installed should work fine if
siphones on the same LAN and subnet work? I forwarded port 5060 to the SME and the
phones still work.

If you have any other suggestions I would appreciate it. What can I do to make helping me easier?

If Wifi has time to make his howto it might solve all problems at once... I hope

Thanks in advance for all the help.

kind regards,

Jan

[wifi]

In SAIL PBX, we 1st need to create a new Carrier.
So select Carriers > New Carrier.

Enter the following in the "Add or Modify a Carrier, Sibling or Gateway" window.

Technology:   SIP
Carrier type:   VOIP
MD5 Encryption: NO
Carrier Name:   BudgetPhone
Description: Dutch Carrier
Host URL:   sip.budgetphone.nl
Registration Template (Optional): username@budgetphone.nl:password:username@sip.budgetphone.nl/username

Now click Save.

Once created, open the just created carrier (BudgetPhone), and add the following to the [peer] box.
fromdomain=budgetphone.nl

The content shown in the peer box should now look like this

[peer]
type=peer
host=
qualify=3000
canreinvite=no
username=
fromuser=
secret=
fromdomain=budgetphone.nl

Click Save again, and than Commit.

Now we need to create the Trunk.
Click on Trunklines, and select New Trunk, and as Carrier the just created Carrier (in my case Budgetphone)
In the DID Number or IP Name, you enter your Budgetphone nr.
REMEMBER. the 31 needs to be added. (example: 31307123456)
Leave the Peer Stanza Label empty, The hostname or IP address should show sip.budgetphone.nl
In the user name enter your Budgetphone nr. Again. Make sure its with the 31 in front. (example: 31307123456)
Enter you Budgetphone password (it's not the same as the forum or login password for the website)
Once created, select the edit option for the Trunk, as you will need to set some things here.
Select the LAN phone nr. that you would like to use for this account. This can be done in the "Open Inbound Route"
As "Transformation Mask" I have the following: 00: 0:31 :00
For me the "Closed Inbound Route" is set to the same LAN phone nr as the "Open Inbound Route"
The "Cluster" is set to default.
The registration string (and this is important) should look like this:
31307123456@budgetphone.nl:2gHd7YtF:31307123456@sip.budgetphone.nl/31307123456
Note:
The 31307123456 should be your number.
The 2gHd7YtF should be your password.

The [31307123456] (again this should be your Budgetphone nr. should look like this)

type=peer
host=sip.budgetphone.nl
qualify=3000
canreinvite=no
username=31307123456
fromuser=31307123456
secret=2gHd7YtF
fromdomain=budgetphone.nl
disallow=all
allow=gsm
allow=alaw
insecure=very

I've done some other tweaks to my SAIL setup, but can not really remember what.
Let me know if you get stuck, and I will try to help you more.

[jreijsenbach]

Thanks for the howto Wifi!

Cannot check if it works today hope to on thursday or friday at the latest. Can't wait to try ...... 

With kind regards,

Jan