Debugging
My Asterisk Config? There are several things
you can do to determine if there is an issue with your configuration.
There are a couple of common mistakes and easy fixes.
One-way Audio
Telasip attempts to support re-invites. Re-invites allow the
audio to be redirected from your box to our providers gatewaey.
Typically this provides the best overall call quality. But
Asterisk needs to be configured for re-invites.
To support re-invites Asterisk needs to know what it's public
IP address is. There are two mechanisms to configure Asterisk
for reinvites. For users with static IP addresses simply at
the following in the sip.conf file in the general section.
externip=xxx.xxx.xxx.xxx ; where xxx.xxx.xxx.xxx is the static
IP address
localnet=xxx.xxx.xxx.xxx/255.255.255.0 ; where xxx.xxx.xxx.xxx
is the internal subnet.
The other mechanism requires you to use Dynamin DNS.
externhost=<yourdynamicdns>
externrefresh=15
localnet=xxx.xxx.xxx.xxx/255.255.255.0
When using Dyanmic DNS make sure Asterisk is using an external
DNS server, and the hostname resolves to the public IP address.
Inbound Calls go straight to voicemail
This is typically an issue with the inbound route. The context=
setting in the peer controls which context the inbound call
will be routed. You need to have an entry for the 10 Digit
Number in that context. To debug how the inbound call is being
routed do the following:
Login in to the linux box
Run asterisk -r to get into asterisk
CLI> sip debug
CLI> set verbose 4
Place inbound call. In the output you will find the text
"Looking for [NXXNXXXXXX] in [context]". This will
tell you how the inbound call is being routed.
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