Hi all,
I’m new to Asterisk and would appreciate some assistance. Do let me know if I must provide any other info and I’d be happy to do that.
My Setup: I’m using Virtualbox installed on my Mac. I have UbuntuVMs on the VirtualBox.
- Asterisk Server on UbuntuVM1
- Linphone on UbuntuVM2
Both VMs have Host Only adapter + NAT adapter enabled so that they can all access internet as well as ping each other (side note: Bridged adapter has trouble with connecting to WiFi on Mac and hence why I’m not using the bridged adapter instead).
Now, static IP addresses have been assigned to both VMs.
Asterisk Server - 192.168.56.103
Linphone on Ubuntu - 192.168.56.104
/etc/asterisk/extensions.conf
[public] exten => 30,1,Answer() exten => 30,n,Background(/var/lib/asterisk/sounds/ivr_promt_user) ;exten => 30,n,SayDigits(1234)
[phones]
exten => 30,1,Answer() exten => 30,n,Background(/var/lib/asterisk/sounds/ivr_promt_user) ;exten => 30,n,SayDigits(1234)
/etc/asterisk/sip.conf
[general] bindaddr=0.0.0.0 context=public host=dynamic type=friend allow=ulaw,alaw
[steve] bindaddr=0.0.0.0 type=friend context=phones allow=ulaw,alaw secret=12345678 host=dynamic
[clive] bindaddr=0.0.0.0 type=friend context=phones allow=ulaw,alaw secret=87654321 host=dynamic
I started asterisk server (on first VM) using sudo asterisk -rvvvvv.
Then I started linphone on my second VM. Clicked on Options > Preferences > Wizard > “I have already a sip account and I just want to use it” > Enter Username: steve, Password: 12345678, Domain: 192.168.56.103 (this is the ip address of the VM that Asterisk is on).
Asterisk CLI says:
--Registered SIP 'steve' at 192.168.56.104:5060 > Saved useragent "Linphone/3.9.1 (belle-sip/1.4.2)" for peer steve
Now, I try calling sip:30@192.168.56.103 and hear nothing on the linphone side (the call gets answered, there is silence and then the call hangs up).
Asterisk CLI says:
== Using SIP RTP CoS mark 5 -- Executing [30@phones:1] Answer("SIP/steve-00000000", " ") in new stack -- Executing [30@phones:2] BackGround("SIP/seteve-00000000", "/var/lib/asterisk/sounds/ivr_promt_user") in new stack -- <SIP/steve-00000000> Playing 'var/lib/asterisk/sounds/ivr_promt_user.ulaw' (language 'en') > 0x7f037000d4f0 -- Probation passed - setting RTP source address to 192.168.56.104:7078 -- Executing [30@phones:3] Hangup("SIP/steve-00000000", " ") in new stack == Spawn extension (phones, 30, 3) exited non-zero on 'SIP/steve-00000000'
Now, if I delete the user account from Linphone (Options > Preferences > Select sip:steve@192.168.56.103 from under Account section > click Remove),
Asterisk CLI says:
--Unregistered SIP 'steve'
Now, on Linphone, my default settings are as follows:
Your display name:
Your username: steve
Your resulting SIP address: ""sip:steve@10.0.3.15:5060 (note that 10.0.3.15 is the IP address for this VM because of the NAT adapter. This field is not editable)
Now if I call sip:30@192.168.56.103, it works - i.e. the call is answered, I hear the IVR message, and the call hangs up.
Asterisk CLI says:
== Using SIP RTP CoS mark 5 -- Executing [30@phones:1] Answer("SIP/steve-00000001", " ") in new stack -- Executing [30@phones:2] BackGround("SIP/seteve-00000001", "/var/lib/asterisk/sounds/ivr_promt_user") in new stack -- <SIP/steve-00000001> Playing 'var/lib/asterisk/sounds/ivr_promt_user.ulaw' (language 'en') > 0x7f037000d4f0 -- Probation passed - setting RTP source address to 192.168.56.104:7078 -- Executing [30@phones:3] Hangup("SIP/steve-00000001", " ") in new stack == Spawn extension (phones, 30, 3) exited non-zero on 'SIP/steve-00000001'
Would anybody help me understand this / overcome this problem?