Hi Guys,
I’m trying to reconfigure my asterisk server to use IP Authentication. It was user SIP Registration before but due to moving to a static IP address it now needs to be re-configured as we’ve upgraded our account. I’ve made the below changes, but when i dial the number i get a skipping ring tone, can anyone shed any light?
Extensions.conf
[others]
[from-voip-provider]
exten => 441453XXXXXX,1Dial(SIP/01453704XXX@ext-sip-account,Fg)
exten => 441453XXXXXX,2Hangup()
Sip.conf
[general]
context=others
allowoverlap=no
port=5060
bindaddr=0.0.0.0
srvlookup=yes
externip="myipaddress"
localnet=192.168.0.2/255.255.255.255
allowexternal=yes
allowguest=yes
[ext-sip-account]
type=friend
fromuser="myipaddress"
host="siphostIP"
dtmfmode=rfc2833
port=5060
disallow=all
allow=ulaw
allow=alaw
insecure=very
nat=yes
What do you mean by “skipping ring tone”. It doesn’t sound like a symptom of an authentication problem to me.
Well a ring tone is normally steading, even pulses. When I dial the number it’s very irregular. Also im not seeing anything in the asterisk logs when i dial the number, no failures or anything
Do you mean ring back tone. The actual cadence depends on the phone, or other source, and tends to vary between countries.
If it is coming from the phone, you should certainly have something on the logs, if you set verbose to at least 3. Also, if it is coming from the network, which sounds more likely, you would have something on the log.
If it is broken and from the network, this indicates network overloading, or running Asterisk on a VM with competing VMs.
Looking at it I suspect that my SIP trunk hasn’t been provisioned correctly. Wrong IP address, if this were the case would this possibly present the lack of info in the logs and odd ring tone? If the IP address is then set correctly, should the above configurations work?