Which satellite system?
I think you need some specialized support, even special hardware. Check
out
http://www.groundcontrol.com/igvoip_001.htm
You may need to replace TCP/IP
http://www.mentat.com/skyx/skyx-gateway.html
Paul
Paul Mahler
pmahler@signate.com
phone: 650-207-9855
fax: 877-408-0105
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Olaf Menzel
Sent: Saturday, October 11, 2003 2:42 PM
To: asterisk-users@lists.digium.com; esa.momosat@horz.de;
esa.momosat@dialogis.de; Karl Jonas
Subject: [Asterisk-Users] SIP / IAX over satellite
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@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@OfficeAsteriskIPAddr)
to
exten =>1000,1,Dial(IAX/office@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@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@fokus.fhg.de Internet: http://www.fokus.fhg.de/satcom
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users