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: fokus.fhg.de/satcom
Which satellite system? I think you need some specialized support, even special hardware. Check out groundcontrol.com/igvoip_001.htm You may need to replace TCP/IP 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: fokus.fhg.de/satcom _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com lists.digium.com/mailman/listinfo/asterisk-users