robertlaferla@comcast.net
2005-Dec-04 14:19 UTC
[Asterisk-Users] Getting started with Asterisk and Aastra 9133i
I have a Aastra 9133i phone and would like to do a simple test to make sure everything works. I already assigned an IP address to the phone (I'm able to ping it.) I have Asterisk running (installed Asterisk and Zaptel only) but not configured. I don't have a FXS/FXO card yet but I would like to test out the phone. Ideally, I'd like to be able to setup a mailbox, record a mailbox greeting, and play it back. How do I do this? I ran "make samples" to install the basic config files then: I added this to the sip.conf: [aastra] type=friend host=192.168.0.99 mailbox=1234@default I also added this to the extensions.conf file: Exten => 1234,1,Wait(2) Exten => 1234,2,Record(/tmp/asterisk-recording:gsm) Exten => 1234,3,Wait(2) Exten => 1234,4,Playback (/tmp/asterisk-recording) Exten => 1234,5,wait(2) Exten => 1234,6,Hangup When I dial "1234", nothing happens. I'm not sure if that's how it supposed to work or what.
Robert La Ferla
2005-Dec-05 09:15 UTC
[Asterisk-Users] Getting started with Asterisk and Aastra 9133i
Let me simplify my problem. I have a single Aastra 9133i SIP phone and latest Asterisk from SVN source running on Fedora Core 4. The phone currently says "No Service" I would like to be able to dial "1234" from the phone and get Asterisk to play back an audio message or say some digits. I can't get this to work with either SayDigits or Playback. Please help. =====sip.conf ===== [general] port = 5060 bindaddr = 0.0.0.0 context=tutorial [3006] type=friend username=3006 secret=mypassword host=dynamic canreinvite=no permit=192.168.0.0/24 allow=all mailbox=3006 ==========extensions.conf ========== [tutorial] exten => 1234,1,Answer exten => 1234,2,SayDigits(123456789) ** TFTP directory ** ============mymacaddress.cfg ============ sip line1 auth name: 3006 sip line1 password: mypassword sip line1 user name: 3006 sip line1 display name: "myname" sip line1 screen name: "myname" ======aastra.cfg ====== dhcp: 1 # DHCP enabled. sip silence suppression: 2 # "0" = off, "1" = on, "2" = default sip proxy port: 5060 # 5060 is set by default. sip registrar ip: 192.168.0.99 # IP of registrar. --- THIS IS THE IP of my Asterisk and tftp server sip registrar port: 5060 # 5060 is set by default. sip digit time out: 6 time server disabled: 0 # Time server disabled. time server1: 192.168.0.99 # Enable time server and enter at