[Asterisk-Users] SIP / IAX over satellite

[Olaf Menzel]

Hi all,
------
I tried to use * over satellite, but all my effort did not succeed.

The Asterisk is behind the VSAT and is resposibel for alle the SIP 
clients in a field location.
The clients are notebooks and PDA's running SJPhoen for Windows and 
PocketPC. Unfortunately
I could not find any Linux Client wich worked satisfying. SJ LAbs 
promised a Linux Version at the end of
August but they forget to to publish the year, but this is a offline topic.

In a first approach  I plugged a Snom100 into the network at the 
satellite hub station. This should simulate
a  operator telephone  in the head quarter.The Snom100 should reach all 
the clients in the field network behind the VSAT
and vice versa.

After configuring and rebooting the Snom100 it tried to register with 
the Asterisk, but the registration timed out. Registering
the Snom in the field network was straight forward without any problems.

In a second approach I set up a second Asterisk in the head quarters 
network. The Snom registered with the Asterisk and I could
dial the Snom via the console dial command. I registered a second snom 
in the field network behind the satellite and it registered with
the Asterisk in the field network. Then I tiried to dial the Snom in the 
Ofiice (Head Quarter)  over both Asterisk' but it failed.
The field asterisk responded with a "not found" message from the ofiice 
asterisk.

The last approach was a trial with IAX routing and call transfer and it 
failed as well.

To clarify the situation a little bit you must know the satellite link 
has a propoagation delay of more than 500 ms.

Here my configurations:

approach 1:
-------------

sip.conf
======

;
; SIP Configuration for Asterisk
;
[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind to
context = default               ; Default for incoming calls
disallow=all
allow=ulaw
allow=alaw
tos=lowdelay
;tos=184
maxexpirey=18000                ; Max length of incoming registration we 
allow
defaultexpirey=12000            ; Default length of incoming/outoing 
registration
;notifymimetype=text/plain      ; Allow overriding of mime type in NOTIFY


[opoffice]
type=friend
secret=opoffice
host=dynamic
dtmfmode=rfc2833
mailbox=1000
context=local
callerid="Operator Office" <1000>

[opfield]
type=friend
secret=opfield
host=dynamic
dtmfmode=rfc2833
mailbox=2000
context=local
callerid="Operator Field" <2000>


extensions.conf
===========

;
; Static extension configuration files, used by
; the pbx_config module.
;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified.  Remember that all comments
; made in the file will be lost when that happen
s.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no

;
; The "Globals" category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental 
variabl
e
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=/dev/dsp

[voicemail]
exten => 7,1,Ringing
exten => 7,2,Wait(2)
exten => 7,3,VoicemailMain

[local]
include => voicemail


; SIP Phone Operator Office
exten => 1000,1,Dial,SIP/opoffice|30
exten => 1000,2,Voicemail,u1000
exten => 1000,102,Voicemail,b1000

; SIP Phone Operator Field
exten => 2000,1,Dial,SIP/opfield|30
exten => 2000,2,Voicemail,u2000
exten => 2000,102,Voicemail,b2000


approach 2:
========
I moved the opoffice Snom to the office side and moved the 
configurations for this phone
in  /etc/sip.conf and /etc/extensions.conf to the Asterisk at the office 
side.

Then I added the following line to /etc/extensions.conf to the local 
context:


[local]

include => voicemail

exten =>1000.,1,Dial(SIP/sip:1000 at OfficeAsteriskIPAddr)

; SIP Phone Operator Field
exten => 2000,1,Dial,SIP/opfield|30
exten => 2000,2,Voicemail,u2000
exten => 2000,102,Voicemail,b2000


Then I tried to dial the extension 1000 from the Snom phone in at the 
field location ...


approach 3 
========
Asterisk Field:
---------------

/etc/iax.conf
--------------

[office]
type=friend
host=192.168.1.1 (example)
context=local
allow=all

/etc/extensions.conf
---------------------
I have changed:

exten =>1000,1,Dial(SIP/sip:1000 at OfficeAsteriskIPAddr)

to
exten =>1000,1,Dial(IAX/office at OfficeAsteriskIPAddr/1000)


Then I dialed the extension 1000  at the field snom again and I hoped 
the call would be routed
to the extension 1000 at the offcie asterisk and the snom their will 
ring. But nothing happened.
Do I need a special [inbound] context for the icoming IAX call at the 
Office Asterisk ?

----------------------------------------------------------------------

Conclusions:
-------------
I am not very familiar with IAX routing but what would be the best 
solution for this issue
when I am not able to register over satellite. I want to registser at 
the local Asterisks and
only want to send the Voip (RTP) traffic over satellite. I SIP I can 
dial any user without
remote registration. Why can't I just reach the registered snom phone by 
just dialing his
sip address (sip:1000 at OfficeAsteriskIPAddr) ? Any suggestions ??

regards

Olaf

-- 
Dipl. Ing. Olaf Menzel - System Engineer
FOKUS - Fraunhofer Institute for Open Communication Systems:
- Competence Center for Advanced Satellite Communication
Schloss Birlinghoven, 53754 Sankt Augustin, Germany
Phone: +49-2241-14-3494 Mobile: +49-175-2616161 Fax:+49-2241-14-43494
email: olaf.menzel at fokus.fhg.de Internet: http://www.fokus.fhg.de/satcom